Combined Queue Management and Scheduling Mechanism to Improve Intra-User Multi-Flow QoS in a Beyond 3,5G Network

Author(s):  
Amine Berqia ◽  
Mohamed Hanini ◽  
Abdelkrim Haqiq

Packet scheduling and buffer management are the two important functions adopted in networks design to ensure the Quality of Service (QoS) when different types of packets with different needs of quality share the same network resources. The Packet scheduling policy determines packet service priorities at the output link, it can reduce packet delay and delay jitter for high-priority traffic. The buffer management involves packet dropping and buffer allocation. The overall goal of such schemes proposed in High Speed Downlink Packet Access (HSDPA) is to take advantage of the channel variations between users and preferably schedule transmissions to a user when the channel conditions are advantageous; it does not take in consideration the characteristics of the flows composing the transmitted traffic to the user. This paper compares two queue management mechanisms with thresholds applied for packets transmitted to an end user in HSDPA network. Those mechanisms are used to manage access packets in the queue giving priority to the Real Time (RT) packets and avoiding the Non Real Time (NRT) packets loss. The authors show that the performance parameters of RT packets are similar in the two mechanisms, where as the second mechanism improves the performance parameters of the NRT packets.

2019 ◽  
Vol 1 (1) ◽  
pp. 1-8
Author(s):  
Pradeep S

In currently, the revolution in a high-speed broadband network is the requirement and also endless demand for high data rate and mobility. To achieve above requirement, the 3rd Generation Partnership Project (3GPP) has been established the Long Time Evolution (LTE). LTE has established an improved LTE radio interface named LTE-Advanced (LTE-A) and it is a promising technology for providing broadband, mobile Internet access. But, better Quality of Service (QoS) to provide for customers is the main issue in LTE-A. To reduce the above issue, the packets should be utilized by using one of the most significant function of packet scheduling to upgrading system performance via determines the throughput performance. In existing scheme, the user with poor Channel Quality Indicator (CQI) has smaller throughput issue is not focused. In this paper, a Hybrid Weighted Round Robin with Shortest Job First (HWRR-SJF) Scheduling technique is proposed to enhance efficient throughput and fairness in LTE system for stationary and mobile users. In this proposed scheduling, to schedule users according to a different criterion like fairness and CQI. HWRR-SJF Scheduling has been proposed for scheduling of the users and it produces increased throughput for various SNR values simulated alongside Pedestrian and Vehicular moving models. The proposed method also uses a 4G-LTE filter or Digital Dividend (DD) in order to align the incoming signal. The digital dividend is used to remove white spaces, which refer to frequencies assigned to a broadcasting service but not used locally. The proposed model is very effective for users in terms of the performance metrics like packet loss, throughput, packet delay, spectral efficiency, fairness and it has been verified through MATLAB simulations.


2015 ◽  
Vol 61 (4) ◽  
pp. 409-414 ◽  
Author(s):  
Mohammed Mahfoudi ◽  
Moulhime El Bekkali ◽  
Abdellah Najd ◽  
M. El Ghazi ◽  
Said Mazer

Abstract The Third Generation Partnership Project (3GPP) has developed a new cellular standard based packet switching allowing high data rate, 100 Mbps in Downlink and 50 Mbps in Uplink, and having the flexibility to be used in different bandwidths ranging from 1.4 MHz up to 20 MHz, this standard is termed LTE (Long Term Evolution). Radio Resource Management (RRM) procedure is one of the key design roles for improving LTE system performance, Packet scheduling is one of the RRM mechanisms and it is responsible for radio resources allocation, However, Scheduling algorithms are not defined in 3GPP specifications. Therefore, it gets a track interests for researchers. In this paper we proposed a new LTE scheduling algorithm and we compared its performances with other well known algorithms such as Proportional Fairness (PF), Modified Largest Weighted Delay First (MLWDF), and Exponential Proportional Fairness (EXPPF) in downlink direction. The simulation results shows that the proposed scheduler satisfies the quality of service (QoS) requirements of the real-time traffic in terms of packet loss ratio (PLR), average throughput and packet delay. This paper also discusses the key issues of scheduling algorithms to be considered in future traffic requirements.


Sensors ◽  
2021 ◽  
Vol 21 (17) ◽  
pp. 5763
Author(s):  
Mohammed Amin Lamri ◽  
Albert Abilov ◽  
Danil Vasiliev ◽  
Irina Kaisina ◽  
Anatoli Nistyuk

Because of the specific characteristics of Unmanned Aerial Vehicle (UAV) networks and real-time applications, the trade-off between delay and reliability imposes problems for streaming video. Buffer management and drop packets policies play a critical role in the final quality of the video received by the end station. In this paper, we present a reactive buffer management algorithm, called Multi-Source Application Layer Automatic Repeat Request (MS-AL-ARQ), for a real-time non-interactive video streaming system installed on a standalone UAV network. This algorithm implements a selective-repeat ARQ model for a multi-source download scenario using a shared buffer for packet reordering, packet recovery, and measurement of Quality of Service (QoS) metrics (packet loss rate, delay and, delay jitter). The proposed algorithm MS-AL-ARQ will be injected on the application layer to alleviate packet loss due to wireless interference and collision while the destination node (base station) receives video data in real-time from different transmitters at the same time. Moreover, it will identify and detect packet loss events for each data flow and send Negative-Acknowledgments (NACKs) if packets were lost. Additionally, the one-way packet delay, jitter, and packet loss ratio will be calculated for each data flow to investigate the performances of the algorithm for different numbers of nodes under different network conditions. We show that the presented algorithm improves the QoS of the video data received under the worst network connection conditions. Furthermore, some congestion issues during deep analyses of the algorithm’s performances have been identified and explained.


2014 ◽  
Vol 13 (9) ◽  
pp. 4792-4798 ◽  
Author(s):  
Dr. Anu Chaudhary ◽  
Dr. Satya Prakash Singh

To evaluate and enhance the performance of the High-Speed Data networks some study on network technology is required to be done, it also require to simulate and verify the results and suggests the better solutions for High-Speed Data networks like Real-time Data network (Multimedia data, Voice data). Multiprotocol Label Switching (MPLS) is an emerging technology and plays an important role in the next generation networks by providing Quality of service (QoS). It overcomes the limitations like excessive delays and high packet loss of IP networks by providing scalability and congestion control. The key feature of MPLS is its Traffic Engineering (TE) which is used for effectively measuring the performance of the networks and efficient utilization of network resources. MPLS provides lower network delay, efficient forwarding mechanism, scalability and predictable performance of the services which makes it more suitable for implementing real-time applications such as Voice and video. In this paper performance of Voice over Internet Protocol (VoIP) application is implemented in MPLS network and conventional Internet Protocol (IP) network. NS-2 (Network Simulater-2) is used to simulate the both networks and the comparison is made based on the metrics such as Voice packet delay, voice packet lost probability, throughput, voice packet send and received. The simulation results are analyzed and it shows that MPLS based solution provides better performance in implementing the VoIP application.The simulated results in the form of animations and graphs can be helpful for network operators or designers to determine the number of VoIP calls, to estimate the packet lose probability and other QoS parameters that can be estimated and evaluated for a given network by using NS-2 simulator.


1995 ◽  
Author(s):  
Rod Clark ◽  
John Karpinsky ◽  
Gregg Borek ◽  
Eric Johnson
Keyword(s):  

Author(s):  
Kenneth Krieg ◽  
Richard Qi ◽  
Douglas Thomson ◽  
Greg Bridges

Abstract A contact probing system for surface imaging and real-time signal measurement of deep sub-micron integrated circuits is discussed. The probe fits on a standard probe-station and utilizes a conductive atomic force microscope tip to rapidly measure the surface topography and acquire real-time highfrequency signals from features as small as 0.18 micron. The micromachined probe structure minimizes parasitic coupling and the probe achieves a bandwidth greater than 3 GHz, with a capacitive loading of less than 120 fF. High-resolution images of submicron structures and waveforms acquired from high-speed devices are presented.


2007 ◽  
Author(s):  
R. E. Crosbie ◽  
J. J. Zenor ◽  
R. Bednar ◽  
D. Word ◽  
N. G. Hingorani

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