An improved queuing model for packet retransmission policy and variable latency decoders

Darabkh, Khalid A.; Jafar, Iyad F.; Sukkar, Ghazi Al; Abandah, Gheith A.; Al‐Zubi, Raed T.

doi:10.1049/iet-com.2012.0410

Cited by 32 publications

(8 citation statements)

References 16 publications

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“…It is quite interesting to notice the trend of approaching infinity for the expected number of awaiting frames as the frame arriving probability steps towards a certain limit that mainly stabilises the decoding system. This limit has been proven to be theoretically existed and is defined as the ratio of one to the expected frame decoding time bearing in mind that the case describes reaching infinite capacity validates our assumption; that is, the maximum allowable capacity is infinite.…”

Section: Analytical Results and Discussionmentioning

confidence: 99%

“…On the contrary, if this frame is so difficult, that is, needs more than provided T time slots, then it is considered corrupted and on that account abandons the system at the end of the T th time slots, thus a decoding failure results. Actually, a partially decoded frame may either subject to an automatic repeat request call to trigger its retransmission or a use of an outer code, such as the Reed–Solomon code, for further error correction .…”

Section: The Proposed Queuing Modelmentioning

confidence: 99%

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Fast and upper bounded Fano decoding algorithm: queuing analysis

Darabkh

2015

Trans. Emerging Tel. Tech.

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In computer networking, having limited buffer size is one of the vital causes of splitting the messages into smaller frames to be then transmitted over the network. Increasing the buffer size will not improve the case much but rather it is likely to degrade the performance of the system if it is not investigated properly. On the other hand, frames' retransmissions, as a result of frames being corrupted or dropped, need to be avoided as much as possible because of possible network congestion. To investigate the problem of undesired retransmissions, this paper addresses the use of convolutional codes utilising variable complexity decoding algorithms, such as the Fano algorithm, which mainly aim to mitigate frames' retransmissions caused by frames' corruption. It is noteworthy to mention that the rapid growth of wireless technologies has attracted everybody's attention. As wireless technology evolves, new decoding algorithms are emerging to demonstrate designs with variable power consumptions. Hence, in this paper, we propose a novel queuing model that describes not only frame arriving and departing processes after getting decoded according to how worse the channel condition is or what is the remaining system power but also a decoder with a double speed compared with what is best scrutinised in the literature. Moreover, in our buffering model, we can reduce the number of frames being dropped because of insufficient buffer size, and unwanted retransmissions owing to long frame waiting time, by virtue of bestowing larger buffer size, through our derived closed-form expression that represents the expected number of awaiting frames in the buffers of Fano decoders parameterised by channel status, decoding threshold and frame arriving probability. As far as the system occupancy is concerned and over what have been previously proposed, our analytical results show impressive improvements. As such, through our proposed queuing model, the required decoder's buffer is substantially smaller of what that obtained by a prior model, considering the same networking parameters, which prompts decreasing the average frame waiting time. Copyright

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Section: Analytical Results and Discussionmentioning

confidence: 99%

Section: The Proposed Queuing Modelmentioning

confidence: 99%

Fast and upper bounded Fano decoding algorithm: queuing analysis

Darabkh

2015

Trans. Emerging Tel. Tech.

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“…Actually, the retransmission queue length is mostly used as an important parameter for loss detection and managing retransmissions [32,33]. Motivated by these facts, this section concentrates on improving the performance of SCTP-based multimedia streaming by constructing a simple, yet effective loss detection and recovery model which is dubbed as RAF-rtx.…”

Section: Algorithm 1 Sctp-rev + -Based Path Selection and Handover Opmentioning

confidence: 99%

Receiver-assisted cellular/wifi handover management for efficient multipath multimedia delivery in heterogeneous wireless networks

Cao

Liu

Zuo³

et al. 2016

J Wireless Com Network

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In a heterogeneous wireless environment, mobility particularly vertical handover between WiFi and cellular data service has to be supported. The support for multihoming feature makes Stream Control Transmission Protocol (SCTP) become a promising transport layer-based handover scheme for end users. Although the benefits of applying SCTP for multimedia delivery and handover have been proved to be extremely useful, there are still many challenges that remain: (i) current sender-centric SCTP handover management solutions fail to balance the computational overhead between a sender and receiver and (ii) they also suffer from performance degradation due to the rigid three-duplicated-SACK-based loss detection and recovery strategy. This paper proposes SCTP-Rev + , a new receiver-assisted cellular/WiFi handover management mechanism for SCTP-based multimedia transport aiming to (i) optimize SCTP handover management and achieve overhead balancing between the sender and receiver, by moving path estimation and handover decision operations from the sender onto receiver and (ii) improve SCTP loss recovery ability and multimedia delivery performance, by providing SCTP with a simple but effective retransmission-aware loss detection/recovery mechanism. We show that the proposed solution outperforms the current SCTP schemes in terms of goodput performance and multimedia delivery quality.

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“…Computer applications have become an essential part of the educational process in almost every field . Science, mathematics, engineering, computer networking, medical specialties, and even literature are some of the fields which use computer programs in their courses to apply the skills taught theoretically in the class room in a practical way and to create a simulated environment which make the learning process more interactive, realistic, and even efficient compared to the old traditional methods . However, to take advantage of these programs, a basic knowledge of computer software is required by the student along with the physical ability to interact with the machine using input and output devices .…”

Section: Introductionmentioning

confidence: 99%

An efficient speech recognition system for arm‐disabled students based on isolated words

Darabkh

Haddad

Sweidan

et al. 2017

Comp Applic In Engineering

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Over the previous decades, a need has emerged to empower human‐machine communication systems, which are essential to not only perform actions, but also obtain information especially in education applications. Moreover, any communication system has to introduce an efficient and easy way for interaction with a minimum possible error rate. The keyboard, mouse, trackball, touch‐screen, and joystick are all examples of tools which were built to provide mechanical human‐to‐machine interaction. However, a system with the ability to use oral speech, which is the natural form of communication between humans instead of mechanical communication systems, can be more practical for normal students and even a necessity for arm‐disabled students who cannot use their arms to handle traditional education tools like pens and notebooks. In this paper, we present a speech recognition system that allows arm‐disabled students to control computers by voice as a helping tool in the educational process. When a student speaks through a microphone, the speech is divided into isolated words which are compared with a predefined database of huge number of spoken words to find a match. After that, each recognized word is translated into its related tasks which will be performed by the computer like opening a teaching application or renaming a file. The speech recognition process discussed in this paper involves two separate approaches; the first approach is based on double thresholds voice activity detection and improved Mel‐frequency cepstral coefficients (MFCC), while the second approach is based on discrete wavelet transform along with modified MFCC algorithm. Utilizing the best values for all parameters in just mentioned techniques, our proposed system achieved a recognition rate of 98.7% using the first approach, and 98.86% using the second approach of which is better in ratio than the first one but slower in processing which is a critical point for a real time system. Both proposed approaches were compared with other relevant approaches and their recognition rates were noticeably higher.

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An improved queuing model for packet retransmission policy and variable latency decoders

Cited by 32 publications

References 16 publications

Fast and upper bounded Fano decoding algorithm: queuing analysis

Fast and upper bounded Fano decoding algorithm: queuing analysis

Receiver-assisted cellular/wifi handover management for efficient multipath multimedia delivery in heterogeneous wireless networks

An efficient speech recognition system for arm‐disabled students based on isolated words

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