Voice Priority Research Articles

Outage Management and House of Quality to Improve Power Station Performance

The Coal Power Plant from Fast Tracking Project (FTP) owned by PT ABC is built by contractors from China. PT ABC is a company working in the area of the provision of electricity services business . In 2010 PT ABC establish business units new special serving the implementation of the operation & maintenance unit whenever the business units services operation & maintenance . Currently has been implement operation & maintenance four units of steam power plants a large scale with, namely Coal Fired Power Plant Indramayu , Coal-Fired Power Plant Rembang , Coal Fired Power Plant Pacitan And Paiton New Coal Fired Power Plant, The condition of generating equipment is not good enough. Maintenance undertaken is currently inadequate again despite the condition of the equipment age is still young. The maintenance of the power plant is needed so that the power plant can improve the quality of service in order to meet the community's guidance and the challenges of technological development. To improve overhaul planning and preparedness at an optimum level, maintenance management is required using Outage Management, which is a synergistic and sustainable process of Planning, Monitoring, Controlling and Evaluation of Planed Outage and using House of Quality (HOQ). Outage Management is used to optimize power plant maintenance activities by applying the rules of Right Problem, Right Solution, Right Design, Right Implementation. By using House of Quality (HOQ), broadly voice of costumer attempted to be converted directly into technical characteristics of maintenance. By applying it is expected the company will be able to achieve technical characteristics in accordance with the target set. from attribute of quizoner made there is Priority Voice of customer (VOC) requirement to Critical Customer Requirements (CCR) are Safety Base Outage Management on Frame Outage Management on priority 1 with score 97 while Outage Management on score 72

Optimal voice packet classification for enhanced VoIP over priority-enabled networks

This paper proposes a method for optimal classification of voice packets to enhance the quality of voice communications over priority-enabled networks when poor transmission conditions occur. Either high or low priority is assigned to each packet according to the relevance of its payload (voice segment) for the voice intelligibility. Then, in case of constrained networking conditions, by discarding first the voice packets of lower importance, the network always delivers those segments that most contribute to the perceptual quality. The proposed method is based on a dynamic programming optimisation algorithm that finds the optimal subset of m high priority voice segments in each utterance of size n > m. Such optimal subset minimizes the reconstruction distortion over all possible subsets with the same size m (i.e., the distortion incurred by a utterance reconstructed from only m segments). The simulation results show that the proposed method consistently achieves higher mean opinion scores (MOS) in comparison with non-selective packet drop under the same random network loss conditions, yielding better quality of experience (QoE) for the same packet loss rates (PLR). The priority classification algorithm is independent from error concealment methods and distortion metrics used in the optimisation process, which allows generalisation for diverse communication networks and applications.

Fast retry limit adaptation for video distortion/delay control in IEEE 802.11e distributed networks

This paper presents a fast retry limit adaptation method for video streaming applications over IEEE 802.11e distributed networks. The method enables each source to adapt the number of retransmissions associated to each video packet by relating the perceived distortion to the drop probability and the acceptable delay to the expiration time, without asking the destination for feedback distortion/delay information. The resulting framework, which is based on a simplified but accurate evaluation of the network statistics and of the distortion introduced by the loss of a specific packet, provides a closed-form, and hence computationally cheap, estimation of the retry limit. Furthermore, with respect to most of the existing solutions, the proposed strategy accounts for the impact of the higher priority voice access category (AC), in order to improve the reliability of the retry limit adaptation in the presence of contending ACs. The method is validated by a simulation platform including the physical communication chain and the 802.11e medium access control layer, and its performance is compared to that obtained from an existing solution and from the optimum theoretical settings.

Voice Priority Queue Scheduling System Models for VoIP over WLANs

The Voice over Internet Protocol (VoIP) is a delay sensitive traffic due to real-time applications on networks. The assessment of voice flow quality in the VoIP is an essential requirement for technical and commercial motivation. The packets of VoIP streaming may experience drops because of the competition among the different kinds of traffic flow over the network. A VoIP application is also sensitive to delay and requires the voice packets to arrive on time from the sender to the receiver side without any delay over WLAN. The scheduling system model for VoIP traffic is an unresolved problem. In this research paper, the author proposes a new Voice Priority Queue (VPQ) scheduling system models and algorithms for the VoIP over WLANs to solve scheduling issues over IP-based networks. They present new contributions, through the three stages of the VPQ. The VPQ scheduling algorithm is provided as an essential technique in the VoIP communication networks to guarantee the QoS requirements. The design of the VPQ is managed by the limited bandwidth utilization and has been proven to have an efficient performance over WLANs.

Fourth Stage of Voice Priority Queue for VoIP over WLANs

Voice over Internet Protocol (VoIP) is growing rapidly during this decade. VoIP is seen as a short-term and long-term transmission for voice and audio traffic and is moving on Wireless Local Area Networks (WLANs) based on IEEE 802.11 standards. Currently, packet scheduling algorithms like Weighted Fair Queuing (WFQ), was mainly designed to provide the bandwidth reservation. The Strict Priority (SP) is low-cost to maintain the delay sensitive voice traffic. Also, a number of research scheduling solutions have been proposed like General processor sharing (GPS), Deficit Round Robin (DRR), Contention-Aware Temporally fair Scheduling (CATS). Unfortunately, the current scheduling won’t be able to handle the VoIP packets properly and they have drawbacks over real-time applications. The objective of this research is to propose a Fourth Stage of Voice Priority Queue (VPQ) packet scheduling and algorithm to ensure more throughput, fairness and efficient packet scheduling for VoIP performance of queues and traffics. A new scheduler flexible which is capable of satisfying the VoIP traffic flows. Experimental topologies on NS-2 network simulator were analyzed for voice traffic.

The VPQ Scheduler In Access Point For VoIP Over WLAN

The Voice over Internet Protocol (VoIP) application has observed the fastest growth in the world of telecommunication. VoIP is seen as a short-term and long-trem transmission for voice and audio traffic. Meanwhile, VoIP is moving on Wireless Local Area Networks (WLANs) based on IEEE 802.11 standards. Currently, there are many packet scheduling algorithms for real-time transmission over network. Unfortunately, the current scheduling will not be able to handle the VoIP packets with the proper manner and they have some drawbacks over real-time applications. The objective of this research is to propose a new Voice Priority Queue (VPQ) packet scheduling and algorithm to ensure more throughput, fairness and efficient packet scheduling for VoIP performance of queues and traffics. A new scheduler flexible which is capable of satisfying the VoIP traffic flows. Experimental topologies on NS-2 network simulator wereanalyzed for voice traffic. Preliminary results show that this can achieve maximum and more accurate VoIP quality throughput and fairness index in access point for VoIP over WLANs. We verified and validated VPQ an extensive experimental simulation study under various traffic flows over WLANs.

Mobility of Mobile Station in Voice Priority Queue Scheduler for VoIPWLANs

Voice over Internet Protocol (VoIP) is growing rapidly during this decade. VoIP is seen as a short-term and long-trem transmission for voice and audio traffic. Meanwhile, VoIP is moving on Wireless Local Area Networks (WLANs) based on IEEE 802.11 standards. Currently, many packet scheduling algorithms like Weighted Fair Queuing (WFQ), was mainly designed to provide the bandwidth reservation. The Strict Priority (SP) is low-cost to maintain the delay sensitive voice traffic. Also, a number of research scheduling solutions have been proposed like General processor sharing (GPS), Deficit Round Robin (DRR), Contention-Aware Temporally fair Scheduling (CATS). Unfortunately, the current scheduling will not be able to handle the VoIP packets with the proper manner and they have some drawbacks over real-time applications. The objective of this research is to propose a new Voice Priority Queue (VPQ) packet scheduling and algorithm to ensure more throughput, fairness and efficient packet scheduling for VoIP performance of queues and traffics. A new scheduler flexible which is capable of satisfying the VoIP traffic flows. Experimental topologies on NS-2 network simulator were analyzed for voice traffic. Preliminary results show that this can achieve maximum and more accurate VoIP quality throughput and fairness index in mobility of mobile station. We verified and validated VPQ an extensive experimental simulation study under various traffic flows over WLANs.

Comprehensive Structure of Novel Voice Priority Queue Scheduling System Model for VoIP Over WLANs

Voice over Internet Protocol (VoIP) has grown quickly in the world of telecommunication. Wireless Local Area Networks (WLANs) are the most performance assuring technology for wireless networks, and WLANs have facilitated high-rate voice services at low cost and good flexibility. In a voice conversation, each client works as a sender or a receiver depending on the direction of traffic flow over the network. A VoIP application requires high throughput, low packet loss, and a high fairness index over the network. The packets of VoIP streaming may experience drops because of the competition among the different kinds of traffic flow over the network. A VoIP application is also sensitive to delay and requires the voice packets to arrive on time from the sender to the receiver side without any delay over WLAN. The scheduling system model for VoIP traffic is an unresolved problem. The objectives of this paper are to identify scheduler issues. This comprehensive structure of Novel Voice Priority Queue (VPQ) scheduling system model for VoIP over WLAN discusses the essential background of the VPQ schedulers and algorithms. This paper also identifies the importance of the scheduling techniques over WLANs.

A Scalable Architecture Supporting QoS Guarantees Using Traffic Engineering and Policy Based Routing in the Internet

The study of Quality of Service (QoS) has become of great importance since the Internet is used to support a wide variety of new services and applications with its legacy structure. Current Internet architecture is based on the Best Effort (BE) model, which attempts to deliver all traffic as soon as possible within the limits of its abilities, but without any guarantee about throughput, delay, packet loss, etc. We develop a three-layer policy based architecture which can be deployed to control network resources intelligently and support QoS sensi-tive applications such as real-time voice and video streams along with standard applications in the Internet. In order to achieve selected QoS parameter values (e.g. loss, delay and PDV) within the bounds set through SLAs for high priority voice traffic in the Internet, we used traffic engineering techniques and policy based routing supported by Border Gateway Protocol (BGP). Use of prototype and simulations validates function-ality of our architecture.

Fair Delay Optimization-based Resource Allocation Algorithm for Video Traffic over Wireless Multimedia System

The major issue related to the realization of wireless multimedia system is the design of suitable medium access control (MAC) protocol. The design challenge is to maximize the utilization of the limited wireless resources while guaranteeing the various quality of service requirements for all traffic classes especially for the stringent real-time constraint of real time variable bit rate (rt-VBR) video service. In this paper a novel resource allocation algorithm for video traffic is proposed. The proposed allocation algorithm aims to provide fair delay for video packets by minimizing the delay difference among transmitted video packets. At the same time it adaptively controls the allocated resources (bandwidth) for video traffic around the corresponding average bit rate, and has the ability of controlling the quality of service (QoS) offered for video traffic in terms of packet loss probability and average delay. A minimized control overhead of only two bits is needed to increase the utilization efficiency. Simulation results show that the proposed algorithm achieves very high utilization and provides nearly fair delay among video packets. Its efficiency is also investigated under traffic integration condition with voice and data traffic to show that the QoS offered to video traffic does not change in the presence of the highest priority voice traffic while data traffic increases the channel utilization to 98% by using the remaining bandwidth after voice and video traffic while a good QoS is offered to voice and data traffic.

Near optimal fair delay based resource allocation for video traffic over wireless multimedia system

The major issue related to the realization of wireless multimedia system is the design of suitable medium access control (MAC) protocol. The design challenge is to maximize the utilization of the limited wireless resources while guaranteeing the various quality of service requirements for all traffic classes especially for the stringent real-time constraint of real-time variable bit rate (rt-VBR) video service. In this paper a novel resource allocation algorithm for video traffic is proposed. The proposed allocation algorithm aims to provide fair delay for video packets by minimizing the delay difference among transmitted video packets. At the same time it adaptively controls the allocated resources (bandwidth) for video traffic around the corresponding average bit rate, and has the ability of controlling the quality of service (QoS) offered for video traffic in terms of packet loss probability and average delay. A minimized control overhead of only two bits is needed to increase the utilization efficiency. Simulation results show that the proposed algorithm achieves very high utilization and provides nearly fair delay among video packets. Its efficiency is also investigated under traffic integration condition with voice and data traffic to show that the QoS offered to video traffic does not change in the presence of the highest priority voice traffic while data traffic increases the channel utilization to 98% by using the remaining bandwidth after voice and video traffic while a good QoS is offered to voice and data traffic.

On improving channel throughput by restricting data traffic access in multimedia integration over wireless channels

A medium access control (MAC) protocol for integrating voice, videoconferencing and email data packet traffic in a high capacity wireless channel while at the same time ensuring the high priority of video and voice is presented and investigated. By combining our past and new scheduling ideas with an idea recently introduced in the literature, our protocol is shown to excel in comparison with other protocols of the literature. The evaluation of the protocol's performance is conducted via an extensive simulation study and we especially focus on the impact of different types and loads of data traffic on the system performance.

Optimal Resource Allocation and Adaptive Call Admission Control for Voice/Data Integrated Cellular Networks

Resource allocation and call admission control (CAC) are key management functions in future cellular networks, in order to provide multimedia applications to mobiles users with quality of service (QoS) guarantees and efficient resource utilization. In this paper, we propose and analyze a priority based resource sharing scheme for voice/data integrated cellular networks. The unique features of the proposed scheme are that 1) the maximum resource utilization can be achieved, since all the leftover capacity after serving the high priority voice traffic can be utilized by the data traffic; 2) a Markovian model for the proposed scheme is established, which takes account of the complex interaction of voice and data traffic sharing the total resources; 3) optimal CAC parameters for both voice and data calls are determined, from the perspective of minimizing resource requirement and maximizing new call admission rate, respectively; 4) load adaption and bandwidth allocation adjustment policies are proposed for adaptive CAC to cope with traffic load variations in a wireless mobile environment. Numerical results demonstrate that the proposed CAC scheme is able to simultaneously provide satisfactory QoS to both voice and data users and maintain a relatively high resource utilization in a dynamic traffic load environment. The recent measurement-based modeling shows that the Internet data file size follows a lognormal distribution, instead of the exponential distribution used in our analysis. We use computer simulations to demonstrate that the impact of the lognormal distribution can be compensated for by conservatively applying the Markovian analysis results.

Resource management policies in GPRS systems

In this paper we consider the problem of resource management in GSM/GPRS cellular networks offering not only mobile telephony services, but also data services for the wireless access to the Internet. In particular, we investigate channel allocation policies that can provide a good tradeoff between the QoS guaranteed to voice and data services end users, considering three different alternatives, and developing analytical techniques for the assessment of their relative merits. The first channel allocation policy, voice priority, gives priority to voice in the access to radio channels; we show that this policy cannot provide acceptable performance to data services, since when all the channels are busy with voice connections, data services perceive long intervals of service interruption. The second channel allocation policy is called R-reservation; it statically reserves a fixed number of channels to data services, thus drastically improving their performance, but subtracting resources from voice users, even when these are not needed for data, thus inducing an unnecessary performance degradation for voice services. The third channel allocation policy is called dynamic reservation; as the name implies, it dynamically allocates channels to data when necessary, using the information about the queue length of GPRS data units within the base station. A threshold on the queue length is used in order to decide when channels must be allocated to data. Numerical results show that the dynamic reservation channel allocation policy can provide effective performance tradeoffs for data and voice services, with the additional advantage of being easily managed through the setting of the threshold value.

Voice quality control under dynamic encoding algorithms in ATM networks

We consider a voice multiplexer which supports ATM packet voice traffic. This system model also applies to the multiplexing operation carried out at the output queue of each output queueing ATM switch. The voice multiplexer implements an encoding algorithm under which the voice quantization levels are dynamically adjusted according to the congestion status of the multiplexer. Using an embedded voice encoding scheme, a voice packet is encoded into lower and higher priority ATM cells. When high congestion conditions are observed, lower priority voice cells (which contain the least significant bits) are discarded. In previous studies, the voice quality under a dynamic encoding algorithm has been described by the long-term average voice encoding rate (AER ∞). Due to the strong correlation properties exhibited by the packet voice traffic, the voice degradation levels incurred under the employed dynamic voice encoding algorithms are not distributed uniformly among the voice sources. As a result, the encoded voice stream for a source can exhibit severe short-term voice quality degradation, while the long-term AER ∞ level remains satisfactory. To correct this situation, we introduce in this paper a “probabilistic” component in the cell-discarding process. A Voice Degradation Indicator (VDI) is introduced to measure the short-term voice quality degradation experienced by a voice stream. Instead of measuring VDI directly from simulations, a combined simulation-analytical method is developed for its calculation. Through this combined method, the length of the required simulation time is significantly reduced to at least 1 10 or 1 20 of that under pure simulations. Through numerical examples, we identify and compare the system performance under the original and the improved algorithms. Under the same AER ∞ levels, introducing a probabilistic component in the cell discarding process is shown to yield better short-term voice quality behavior as well as to improve the mean cell waiting time performance.

Arpeggio generating system and method

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A priority cutoff flow control scheme for integrated voice-data multiplexers

In this paper, we consider the flow control problem for a movable boundary integrated voice-data multiplexer. We propose a flow control scheme where a decision rule based on the data queue length is employed to cutoff the priority of voice to prevent a data queue buildup. A continuous-time queueing model for the integrated multiplexer is developed. The performance of the flow control scheme is obtained using an efficient computational procedure. A numerical example is presented for illustration.