Journal articles on the topic 'Packet delay'
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1
Chandran, Priya, and Chelpa Lingam. "A Statistical Approach to Adaptive Playout Scheduling in Voice Over Internet Protocol Communication." International Journal of Electrical and Computer Engineering (IJECE) 8, no. 5 (October 1, 2018): 2926. http://dx.doi.org/10.11591/ijece.v8i5.pp2926-2933.
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Factors like network delay, latency and bandwidth significantly affect the quality of communication using Voice over Internet Protocol. The use of jitter buffer at the receiving end compensates the effect of varying network delay up to some extent. But the extra buffer delay given for each packet plays a major role in playing late packets and thereby improving voice quality. As the buffer delay increases packet loss rate decreases, which in general is a very good sign. However, an increase of buffer delay beyond a certain limit affects the interactive quality of voice communication. In this paper, we propose a statistical framework for adaptive playout scheduling of voice packets based on network statistics, packet loss rate and availability of packets in the buffer. Experimental results show that the proposed model allocates optimal buffer delay with the lowest packet loss rate when compared with other algorithms.
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Zhang, Min, and Bin Li. "Performance Analysis of Cognitive Radio Networks with a Two-Part Queue." Open Electrical & Electronic Engineering Journal 9, no. 1 (July 31, 2015): 238–46. http://dx.doi.org/10.2174/1874129001509010238.
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In a cognitive radio network (CRN), a preempted secondary user (SU) is placed in a call level queue to wait for accessing another free channel. If the availability of channels is transparent to SUs, packets will be generated during their waiting time and the performance of the CRN will be influenced by which way to handle these packets. In this paper, the call level queue is departed into two parts, delay queue and discard queue. Here, an analytical model is developed to derive the formulas for both call level performance measures (i.e., call blocking probability) and packet level performance measures (i.e., packet delay, packet loss ratio and throughput). Numerical results show that theoretical models are consistent with simulation results. The major observations include (i) The performances of an SU degrade as the call arrival rate increases. (ii) With the increase of the delay queue length, the SU call blocking probability and packet delay increase, while the packet loss ratio and throughput decrease. (iii) Adopting different delay queue length causes a smaller effect on call blocking probability and throughput than on packet loss ratio and packet delay.
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Lee, Woonghee, Joon Yeop Lee, Hyeontae Joo, and Hwangnam Kim. "An MPTCP-Based Transmission Scheme for Improving the Control Stability of Unmanned Aerial Vehicles." Sensors 21, no. 8 (April 15, 2021): 2791. http://dx.doi.org/10.3390/s21082791.
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Recently, unmanned aerial vehicles (UAVs) have been applied to various applications. In order to perform repetitive and accurate tasks with a UAV, it is more efficient for the operator to perform the tasks through an integrated management program rather than controlling the UAVs one by one through a controller. In this environment, control packets must be reliably delivered to the UAV to perform missions stably. However, wireless communication is at risk of packet loss or packet delay. Typical network communications can respond to situations in which packets are lost by retransmitting lost packets. However, in the case of UAV control, delay due to retransmission is fatal, so control packet loss and delay should not occur. As UAVs move quickly, there is a high risk of accidents if control packets are lost or delayed. In order to stably control a UAV by transmitting control messages, we propose a control packet transmission scheme, ConClone. ConClone replicates control packets and then transmits them over multiple network connections to increase the probability of successful control packet transmission. We implemented ConClone using real equipment, and we verified its performance through experiments and theoretical analysis.
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Volochiy, B., A. Kushyk, Yu Salnyk, V. Onyshchenko, and P. Каzаn. "Method of increasing the efficiency of the switch node of the information communication network for special purpose in the conditions of combat use." Military Technical Collection, no. 26 (June 23, 2022): 3–12. http://dx.doi.org/10.33577/2312-4458.26.2022.3-12.
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The peculiarity of the special purpose communication information network is that its functioning is influenced by the tactical situation. In particular, it determines the intensity of the flow of packets with information about the enemy's moving objects from reconnaissance and signaling complexes to the switching node, and from it to the control point and means of destruction, depending on the probable nature of enemy action. In combat applications, the information network operates simultaneously with fast and slow packet flows.
At the same time, the packet delay in the switching node should not exceed the allowable value. Excessive packet delay results in the loss of relevance of transmitted messages and, consequently, in the loss of intelligence data. The proposed method of eliminating excessive delay involves the formation of two queues of packets in the switching node. Withdrawal of service packages from two queues is carried out using a new adaptive procedure.
An adaptive switch has been introduced into the structure of the switching node, designed to withdraw packets from queues to the packet service system. When selecting a packet queue, the adaptive switch compares the number of service requests from each queue. The method assumes that the number of service requests from the fast packet flow queue determines the number of packets that are in the queue. The number of requests that are formed from the queue of slow-flow packets has two components: the number of real and the number of conditional requests. Actual applications take into account the number of packets in the queue.
Real requests take into account the number of packets in the queue. Conditional requests take into account the delay time, since for each missed cycle of the packet's withdrawal from the slow flow queue, the adaptive switch generates a conditional request. Therefore, the number of requests for a slow stream grows even without packets entering the queue. A comparative study of options for implementing the adaptive procedure has been carried out. The proposed method for eliminating excessive packet delay in the switching node provides an increase in the efficiency of the functioning of a special-purpose information network as a whole.
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Yanto, Rudi, Dedy Irfan, and Asrul Huda. "Analisis Quality of Service Jaringan Wireless untuk Teknologi Streaming." Edumatic: Jurnal Pendidikan Informatika 6, no. 2 (December 20, 2022): 167–75. http://dx.doi.org/10.29408/edumatic.v6i2.5840.
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Wireless network services can be known for their quality with the Quality of Service (QoS) method. This method can measure the quality of streaming services in terms of throughput, delay, packet loss, and jitter. The purpose of this study was to analyze wireless network QoS for streaming technology. This type of research is quantitative with observation methods using the Wireshark application and compared with TIPHON standards. The target of the ICONNET ISP wireless survey on Jalan Nusantara km. 13 Tanjungpinang Timur District. The parameters for measuring QoS use four parameters, namely throughput, delay, packet loss, and jitter. Our findings show that the throughput of obtaining index value is 3.67 and is at a good level. Furthermore, the delay value has an average index of 4 with the best level. Meanwhile, the jitter obtained an index value of 3 at a good level, while the packet loss value obtained an index of 3.3 and had a good level. Based on the results of this study, the quality of ICONNET ISP wireless network services when accessing streaming technology shows data speed instability, data delays, and lost data packets. However, the network quality is still in the "Good" level in terms of throughput, jitter, and packet loss parameters, and the "Best" level in terms of the delay parameter.
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Berqia, Amine, Mohamed Hanini, and Abdelkrim Haqiq. "Combined Queue Management and Scheduling Mechanism to Improve Intra-User Multi-Flow QoS in a Beyond 3,5G Network." International Journal of Mobile Computing and Multimedia Communications 4, no. 1 (January 2012): 57–68. http://dx.doi.org/10.4018/jmcmc.2012010105.
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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.
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Razavi, Rouzbeh, Martin Fleury, and Mohammed Ghanbari. "Adaptive Packet-Level Interleaved FEC for Wireless Priority-Encoded Video Streaming." Advances in Multimedia 2009 (2009): 1–14. http://dx.doi.org/10.1155/2009/982867.
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Packet-level Forward Error Control (FEC) for video streaming over a wireless network has received comparatively limited investigation, because of the delay introduced by the need to assemble a group of packets. However, packet-level interleaving when combined with FEC presents a remedy to time-correlated error bursts, though it can further increase delay if this issue is not addressed. This paper proposes adapting the overall degree of interleaved packet-level FEC according to the display deadlines of packets, transmit buffer occupation, and estimated video input to the wireless channel, all of which address the issue of delay. To guard against estimation error, the scheme applies a conservative adaptation policy, which accounts for picture type importance to ensure that display deadlines are met, thus avoiding this defect of interleaving. The paper additionally introduces a greedy algorithm that effectively groups packet-level FEC protection according to packet priority. Priority encoding adds extra protection during deep fades. As feedback is not required, the interleaving scheme is suitable for all forms of video broadcast. A Bluetooth piconet demonstrates the packet-level FEC interleaving scheme, which provides higher quality delivered video compared to the industry-standard Pro-MPEG Cop#3r2 interleaving scheme.
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Zukerman, Moshe. "Applications of matrix-geometric solutions for queueing performance evaluation of a hybrid switching system." Journal of the Australian Mathematical Society. Series B. Applied Mathematics 31, no. 2 (October 1989): 219–39. http://dx.doi.org/10.1017/s0334270000006603.
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AbstractWe consider a hybrid switch which provides integrated packet (asynchronous) and circuit (isochronous) switching. Queue size and delay distribution of the packet switched traffic in the steady state are derived by modelling the packet queue as a queue in a Markovian environment. The arrival process of the packets as well as of the circuit allocation requests are both modelled by a Poisson process. The analysis is performed for several circuit allocation policies, namely repacking, first-fit (involving static or dynamic renumbering) and best-fit. Both exact results and approximations are discussed. Numerical results are presented to demonstrate the effect of increase in packet and circuit loading on the packet delay for each of the policies.
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Karim, Abdul, and Andi Achmadi. "Analisis Kinerja Koneksi Jaringan Switch Ethernet pada Local Area Network (LAN)." Ainet : Jurnal Informatika 1, no. 1 (August 22, 2019): 1–6. http://dx.doi.org/10.26618/ainet.v1i1.2283.
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The development of today's telecommunication network is progressing very fast. Various kinds of telecommunication technology facilities continue to be developed so that users can communicate practically, wherever the location of the user the resides. This study aims to find out what processes are running when data packets are sent such as computing and comparing the speed of packet data transmitted (Throughput), packet loss data (Packet Loss), and time delay data transmission from each user used. The method used in this study is to use literature study by taking data in the field with the aim to Calculate the value of Throughput, delay and packet loss of a packet sent by each user in the network Ethernet Switch. From the results of the study obtained Comparison of Througput of each field of work, seen in the financial field has the highest value. While having the lowest Througput value is in the field of news. Comparison of time delay of each field can be in the field of delay news with the fastest time to send data and in the field of finance delay with the longest time in data transmission. The packet loss data analysis of each experiment gets a good average score in each field according to ITU G.114 standard with 0% percentage.
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Lamri, Mohammed Amin, Albert Abilov, Danil Vasiliev, Irina Kaisina, and Anatoli Nistyuk. "Application Layer ARQ Algorithm for Real-Time Multi-Source Data Streaming in UAV Networks." Sensors 21, no. 17 (August 27, 2021): 5763. http://dx.doi.org/10.3390/s21175763.
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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.
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Amirsaidov, Ulugbek, and Azamat Qodirov. "A Packet Delay Assessment Model in the Data Link Layer of the LTE." JOIV : International Journal on Informatics Visualization 5, no. 4 (December 28, 2021): 402. http://dx.doi.org/10.30630/joiv.5.4.601.
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The issues of modeling and evaluating the characteristics of the LTE data link layer functioning are considered. Transmitting packets in the data link layer are represented by a probabilistic-temporal graph consisting of two subgraphs. The first subgraph describes the operation of the HARQ protocol, and the second subgraph describes the operation of the ARQ protocol. The first subgraph is nested within the second subgraph. The probabilities of correct reception, non-error detection, and retransmission of packets in the MAC and RLC layers and generating functions of the packet service time based on the HARQ and ARQ protocols are determined. With the help of generating functions, the average value, variance, and coefficient of variation of the packet service time are determined. To calculate the average packet delay time in the LTE data link layer, the type of queuing system is selected, taking into account the coefficient of variation of the packet service time. The analysis of packets' delay time in the network's data link layer is carried out for different values of the intensity of packet arrival and the probabilities of a bit error in the physical layer of the network. For the sustainable functioning of the data link layer of the network, the limit values of the intensity of the arrival of packets are determined for a given probability of a bit error in the physical layer of the network.
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TANG, MINGHUA, and XIAOLA LIN. "RQRT: REDUCE QUERYING ROUTING TABLE FOR MESH-BASED NETWORK-ON-CHIP." Journal of Circuits, Systems and Computers 20, no. 08 (December 2011): 1529–45. http://dx.doi.org/10.1142/s0218126611008018.
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Although using table to implement routing algorithm has some advantages in network-on-chip (NoC), the router queries the routing table whenever a packet is to be forwarded. The querying time significantly increases the packet delay even if some methods have been proposed to shorten the table size. In mesh-based NoC, statistics shows that two neighbor routers have the same routing options for over 50% packets, on average. In this paper, we propose a technique to let packets pass through some routers without querying the routing table. Consequently, the time to query the routing table is significantly decreased. Simulation results show that this leads up to 16% decrease of average packet delay.
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V. Jatti, Ashwini, and Dr V. J. K. Kishor Sonti. "Sinkhole Attack Detection and Prevention using Agent Based Algorithm." Journal of University of Shanghai for Science and Technology 23, no. 05 (May 24, 2021): 526–44. http://dx.doi.org/10.51201/jusst/21/05175.
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This study presents sinkhole attack detection and prevention using agent-based algorithm. In this algorithm, agents are used to provide information to all node from its reliable neighbors by negotiation in three steps, thus nodes may not be able to pay the attention to the traffic made by sinkhole attacker. In this work, network scale of 500×500 m2 square areas have been considered. Series of simulation are carried in each experiment. Every simulation run is being organized to work for 10mins. Network performance is evaluated in terms of throughput, packet delivery ratio, jitter, delay in packets delivery, data packets received, data packets drop using network simulations software. Network simulation results depicts that in proposed algorithm, throughput increases by 15 to 20 percent, packet delivery ratio increases by 30 to 40%, decrease in the jitter by 10 to 15 %, delay in packets delivery is decreased by 15 to 20 %, data packets received are increased by 15 to 20 % and number of the data packets drop are decreased by 5 to 15 %. Based on simulation results throughput, packet delivery ratio and data packets received increased in proposed agent-based algorithm. However, it is observed that, jitter, delay in packets delivery and data packets drop were decreased.
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Hanji, Bhagyashri R., and Rajashree Shettar. "Cross Layer Solution for Energy and Delay Optimization in MANETs." International Journal of Electrical and Computer Engineering (IJECE) 8, no. 6 (December 1, 2018): 4745. http://dx.doi.org/10.11591/ijece.v8i6.pp4745-4754.
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A novel method for packet forwarding in MANETs has been proposed in this paper. A node in the network acts as both host and router. Energy utilization of the node increases as all nodes in MANET operate as source, destination, and router to forward packets to the next hop ultimately to reach destination. Routers execute a variety of functions from simple packet classification for forwarding to complex payload revision. As the number of tasks and complexity increases, processing time required also increases resulting in significant processing delay in routers. The proposed work optimizes packet header at transport and network layer by calculating Unique Identifier using pairing function for the fields which do not change for a source–destination pair. This technique optimizes the processing cost of each packet header thereby conserving energy and reducing delay. It also simplifies the task of system administration. This paper elucidates an extension to basic AODV protocol, allowing routing of most packets without an explicit header, reducing the overhead of the protocol while still conserving its basic properties. The proposed method improves the network performance significantly compared to AODV, MTPR, and S-AODV protocol.
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Hu, Jinyu, Juan Luo, Yuxi Zhang, Panwu Wang, and Yu Liu. "Location-Based Data Aggregation in 6LoWPAN." International Journal of Distributed Sensor Networks 2015 (2015): 1–9. http://dx.doi.org/10.1155/2015/912926.
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Location-based information has recently been exploited to assist the aggregated process of data, thereby reducing the spatial redundancy efficiently. The constraints nature in 6LoWPAN becomes one of the major concerns in data aggregation methods. However, traditional CSMA/CA in MAC layer may cause significant transmission and control overhead as well as delay on listening and competing for channels. It is a low efficient way to transfer IPv6 packet due to the big packet header. To overcome these shortages, in this paper, we propose LDAA, a location-based novel data aggregation model that aggregates data from the network layer according to the MAC layer queuing delay. When the queuing delay becomes larger, more packets will be dynamically aggregated into one packet to increase the proportion of application data. Otherwise, the amount of packets involved in aggregation will decrease to improve channels utilization. Simulation results show that our approach could provide better real-time guarantees and reduce data spatial redundancy and energy consumption efficiently.
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Gohar, Moneeb, Sajid Anwar, Moazam Ali, Jin-Ghoo Choi, Hani Alquhayz, and Seok-Joo Koh. "Partial Bicasting with Buffering for Proxy Mobile IPV6 Mobility Management in CoAP-Based IoT Networks." Electronics 9, no. 4 (March 31, 2020): 598. http://dx.doi.org/10.3390/electronics9040598.
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Constrained application protocol (CoAP) can be used for message delivery in wireless sensor networks. Although CoAP-based proxy mobile internet protocol (PMIP) was proposed for mobility management, it resulted in handover delay and packet loss. Therefore, an enhanced PMIP version 6, with partial bicasting in CoAP-based internet of things (IoT) networks, is proposed. Here, when an IoT device moved into a new network, the corresponding mobile access gateway (MAG) updated the local mobility anchor (LMA) binding. Further, LMA initiated the “partial” bicasting of data packets to the new and the previous MAGs. The data packets were buffered at the new MAG during handover and were forwarded to Mobile Node (MN) after the handover operations. The proposed scheme was compared with the existing scheme, using ns-3 simulations. We demonstrated that the proposed scheme reduced handover delays, packet losses, end-to-end delay, throughput, and energy consumption, compared to the existing scheme.
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Ge, Yuan, Qigong Chen, Ming Jiang, and Yiqing Huang. "Modeling of Random Delays in Networked Control Systems." Journal of Control Science and Engineering 2013 (2013): 1–9. http://dx.doi.org/10.1155/2013/383415.
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In networked control systems (NCSs), the presence of communication networks in control loops causes many imperfections such as random delays, packet losses, multipacket transmission, and packet disordering. In fact, random delays are usually the most important problems and challenges in NCSs because, to some extent, other problems are often caused by random delays. In order to compensate for random delays which may lead to performance degradation and instability of NCSs, it is necessary to establish the mathematical model of random delays before compensation. In this paper, four major delay models are surveyed including constant delay model, mutually independent stochastic delay model, Markov chain model, and hidden Markov model. In each delay model, some promising compensation methods of delays are also addressed.
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Venu, Nookala, D. Yuvaraj, J. Barnabas Paul Glady, Omkar Pattnaik, Gurpreet Singh, Mahesh Singh, and Amsalu Gosu Adigo. "Execution of Multitarget Node Selection Scheme for Target Position Alteration Monitoring in MANET." Wireless Communications and Mobile Computing 2022 (June 8, 2022): 1–9. http://dx.doi.org/10.1155/2022/2088289.
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In mobile network, nodes are normally placed in some locations after travelling with various speeds to another location. Packets were broadcast to some location receiver node, but they are moved to another location, due to that node is not able to receive those packets. Attacker node present in routing path should accept those packets, and it acts as original node. Communication privacy is reduced for mobile network. It improves the communication overhead and end to end delay. So, the proposed Enhanced Packet Acceptance for Target Position Alteration (EPATP) technique exactly monitors the target node position, depending on the position to assign the relay node for packet forwarding from sender to target node. Multiaccepter Assigning Algorithm is designed, and if any target node should not receive those packets, it provides another chance for packet receiving by next target node, and it assigns multiple target node for accuracte communication. It reduces communication overhead and end to end delay.
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Yue, Wuyi, and Yutaka Matsumoto. "Output and Delay Processes in a Slotted ALOHA Multichannel Packet Radio Network with Capture." Probability in the Engineering and Informational Sciences 6, no. 4 (October 1992): 471–93. http://dx.doi.org/10.1017/s0269964800002680.
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In this paper, we exactly analyze the performance of the slotted ALOHA access scheme with capture in a multichannel packet radio communication environment for the IFT (immediate-first-transmission) protocol and the DFT (delayed-first-transmission) protocol. We derive four moment generating functions for the following performance measures: (1) the number of packet deparures in each group of capture level in any slot, (2) the interval time between wo consecutive slot ends with the same number of departures, (3) the interval time between two consecutive slot ends with at least one departure and the number of departures in each group in that slot, and (4) the packet delay for each group. We calculate the averages and higher moments of these performance measures by differentiating the moment generating functions and numerically compare the systems with and without ower capture. The system consists of a finite population of Nstations, both fixed and mobile, that are divided into L different capture groups and access a set of parallel M channels to transmit their packets. Capture effect means that a packet transmitted by a station with a highest capture level can be received accurately, even when other packets in lower capture levels are simultaneously transmitted on the same channel and in the same slot. Numerical comparison to a multichannel system without capture is made. Capture effects on channel utilization, mean packet delay, and coefficients of variation of packet delay and interdeparture time are examined.
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Kaneyasu, T., Y. Hikosaka, M. Fujimoto, H. Iwayama, and M. Katoh. "Coherent control of atoms in the extreme ultraviolet and attosecond regime by synchrotron radiation." Journal of Physics: Conference Series 2380, no. 1 (December 1, 2022): 012115. http://dx.doi.org/10.1088/1742-6596/2380/1/012115.
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Abstract Control of interference between wave packets is one of the basic concepts in coherent control that enables quantum manipulation of populations and reaction pathways in matter. We have recently shown a new method to achieve coherent control in the extreme ultraviolet and attosecond regime using synchrotron radiation. This method is based on the use of longitudinal coherence of light wave packets that are naturally included in the undulator radiation. For quantum manipulation of atomic systems, wave packet interference is precisely controlled by tuning the time delay between the light wave packets. Here we show that the quantum phase controlled by the time delay can be monitored as an initial phase of the quantum beat oscillation in fluorescence decay.
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Ahamed, Afsana, and Hamid Vakilzadian. "Impact of Direction Parameter in Performance of Modified AODV in VANET." Journal of Sensor and Actuator Networks 9, no. 3 (September 3, 2020): 40. http://dx.doi.org/10.3390/jsan9030040.
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A vehicular ad hoc network (VANET) is a technology in which moving cars are used as routers (nodes) to establish a reliable mobile communication network among the vehicles. Some of the drawbacks of the routing protocol, Ad hoc On-Demand Distance Vector (AODV), associated with VANETs are the end-to-end delay and packet loss. We modified the AODV routing protocols to reduce the number of route request (RREQ) and route reply (RREP) messages by adding direction parameters and two-step filtering. The two-step filtering process reduces the number of RREQ and RREP packets, reduces the packet overhead, and helps to select the stable route. In this study, we show the impact of the direction parameter in reducing the end-to-end delay and the packet loss in AODV. The simulation results show a 1.4% reduction in packet loss, an 11% reduction in the end-to-end delay, and an increase in throughput.
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OIDA, K., and K. SHINJO. "CHARACTERISTICS OF DETERMINISTIC OPTIMAL ROUTING FOR TWO HETEROGENEOUS PARALLEL SERVERS." International Journal of Foundations of Computer Science 12, no. 06 (December 2001): 775–90. http://dx.doi.org/10.1142/s0129054101000862.
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This paper presents characteristics of optimal routing that assigns each arriving packet to one of two heterogeneous parallel servers, each with its own queue. The characteristics are derived from numerical solutions to an optimization problem, which is to find optimal routing that minimizes the average packet delay under the condition that all of the packets' arrival times as well as all of the packets' sizes are completely known in advance. There are four characteristics: (1) Under light or moderate traffic, the average packet delay of optimal routing is almost the same as that of join the shortest delay (JSD) policy. (2) Under heavier traffic, optimal routing comes to more often use fix queue based on size (FS) policy. (3) Under heavy traffic, optimal routing assigns small packets to the slower server. (4) As the ratio of the slower server's service rate to the faster server's service rate decreases, optimal routing comes to more often use FS policy under light or moderated traffic. These characteristics are verified by the fact that a mimic optimal routing designed based on the four characteristics attains almost the same performance as optimal routing.
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Pertovt, Erik, Kemal Alic, Aleš Švigelj, and Mihael Mohorcic. "Performance Evaluation of VoIP Codecs over Network Coding in Wireless Mesh Networks." WSEAS TRANSACTIONS ON COMMUNICATIONS 20 (December 28, 2021): 185–91. http://dx.doi.org/10.37394/23204.2021.20.24.
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Voice over Internet protocol (VoIP) is used for transmitting voice signals in a packet-switched Internet protocol (IP) networks in real time. For transmitting voice over a wireless mesh networks (WMNs), the analog voice signal has to be digitalized, encoded, and packetized. Codecs based on different Quality of Service (QoS) requirements are used. One of the main QoS requirements is that packets are transmitted through the network in real time; one-way transmission time or End-to-End (ETE) packet delay, and packet delay variation or jitter have to be lower than thresholds. ETE delay depends on various parameters; among them is also network delay. Various mechanisms are used to lower the network delay in WMNs. A promising mechanism, for improving the performance of streaming services such as the case also in VoIP, is network coding. In this paper, we evaluate the benefits of using wireless network coding for VoIP in WMNs. Network coding procedure in combination with various VoIP codecs is used to observe the impact on network delay and jitter of the VoIP application. The simulation results show that network coding can decrease network delay and jitter. Moreover, results show that network coding benefits are codec dependent.
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Lun, Weicheng, Qun Li, Zhi Zhu, and Can Zhang. "Routing Strategies for Isochronal-Evolution Random Matching Network." Entropy 25, no. 2 (February 16, 2023): 363. http://dx.doi.org/10.3390/e25020363.
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In order to abstract away a network model from some real-world networks, such as navigation satellite networks and mobile call networks, we proposed an Isochronal-Evolution Random Matching Network (IERMN) model. An IERMN is a dynamic network that evolves isochronally and has a collection of edges that are pairwise disjoint at any point in time. We then investigated the traffic dynamics in IERMNs whose main research topic is packet transmission. When a vertex of an IERMN plans a path for a packet, it is permitted to delay the sending of the packet to make the path shorter. We designed a routing decision-making algorithm for vertices based on replanning. Since the IERMN has a specific topology, we developed two suitable routing strategies: the Least Delay Path with Minimum Hop (LDPMH) routing strategy and the Least Hop Path with Minimum Delay (LHPMD) routing strategy. An LDPMH is planned by a binary search tree and an LHPMD is planned by an ordered tree. The simulation results show that the LHPMD routing strategy outperformed the LDPMH routing strategy in terms of the critical packet generation rate, number of delivered packets, packet delivery ratio, and average posterior path lengths.
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D. Gangadhar, Nandyala, and Govind R. Kadambi. "Delay Distributions in Discrete Time Multiclass Tandem Communication Network Models." International journal of electrical and computer engineering systems 13, no. 6 (September 1, 2022): 417–25. http://dx.doi.org/10.32985/ijeces.13.6.1.
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An exact computational algorithm for the solution of a discrete time multiclass tandem network with a primary class and cross-traffic at each queue is developed. A sequence of truncated Lindley recursions is defined at each queue relating the delays experienced by the first packet from consecutive batches of a class at that queue. Using this sequence of recursions, a convolve-and-sweep algorithm is developed to compute the stationary distributions of the delay and inter-departure processes of each class at a queue, delays experienced by a typical packet from the primary class along its path as well as the mean end-to-end delay of such a packet. The proposed approach is designed to handle the non-renewal arrival processes arising in the network. The algorithmic solution is implemented as an abstract class which permits its easy adaptation to analyze different network configurations and sizes. The delays of a packet at different queues are shown to be associated random variables from which it follows that the variance of total delay is lower bounded by the sum of variances of delays at the queues along the path. The developed algorithm and the proposed lower bound on the variance of total delay are validated against simulation for a tandem network of two queues with three classes under different batch size distributions.
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ŞİMŞEK, Mehmet, Nurettin Doğan, and Muhammet Ali Akcayol. "A New Packet Scheduling Algorithm for Real-Time Multimedia Streaming." Network Protocols and Algorithms 9, no. 1-2 (June 30, 2017): 28. http://dx.doi.org/10.5296/npa.v9i1-2.12410.
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Delivering the real-time services over converged networks is a big challenge. Real-time services need to high Quality of Service (QoS). For this purpose, bandwidth reservation and packet prioritization techniques are used. Thus, real-time data packets can be reached to their targets with minimum delays and losses. But, this situation creates unintended consequences for other internet services such as HTTP and FTP. In this case, establishing a balance between the real-time services and the other services is a must. In this study we introduce a new research question: how to transport real-time multimedia IP packets just in time? Just in time means that transportation of the packets neither early, nor late. For this purpose we developed a scheduling/prioritizing algorithm called just in time transport (JITT). Following a cross-layer design approach, JITT controls delay and jitter over whole communication path. We evaluated JITT on the different simulations and one experimental testbed for performance analysis. Our findings support that JITT provides stable delay and low jitter and transports the packets nearly just in time.
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Massey, W. A., and R. Srinivasan. "A packet delay analysis for cellular digital packet data." IEEE Journal on Selected Areas in Communications 15, no. 7 (1997): 1364–72. http://dx.doi.org/10.1109/49.622918.
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Et. al., Dr B. Shadaksharappa,. "Attack Prediction By Using Greedy Algorithm For Diminishing The Drop And Delay In Wireless Sensor Networks." Turkish Journal of Computer and Mathematics Education (TURCOMAT) 12, no. 6 (April 11, 2021): 1072–82. http://dx.doi.org/10.17762/turcomat.v12i6.2425.
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The essential constraint of the internet is that forwarding the data packets of data among the restricted and trustworthy data nodes. If the receiver node is attacker node then it'll drop the data rather than forwarding the data to ensuing neighbor node. Therefore, efficient and secure data transmission is extremely necessary within the network data transmission. Each router node within the network can accept the data packets up to its buffer size only. Once the queue value reached the buffer threshold value then congestion can occur at the node. Once congestion happens then it would lose the data packets. By sending the data packets to the next neighbour node this problem will be resolved. This congestion will be handled by the Fully Distributed Congestion Control FDCC and Cooperative and Memory Efficient Token Bucket (CMTB) algorithms. Because the data is transmitted to the next neighbour node predicting the node behavior is extremely necessary because it is an attacker or the conventional transmitter node because it has to transmit the efficient data securely to the destination node. In this paper, the node behavior will be predicted by analyzing the trace file. The simulation results show that this proposed method would provide a lot of security in data transmission. The WSN comprises a group of sensor nodes that are disseminated on the network. These sensor nodes initially exchange their data packets to the near nodes to send the data packets to the target node. During the transmission of these data packets some data packets drop may also happen inside the network. This packet drop should be kept up as low as feasible for correct data transmission to the target node or destination node. This algorithm highlights the routes with high link quality, low packet delay and with low packet drop. Simulation results show that this proposed algorithm can provide the most effective path for transmitting the data to the destination meanwhile it reduces the packet drop and packet delay.
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Pang, Zhong-Hua, Zhen-Yi Liu, Zhe Dong, and Tong Mu. "An Event-Triggered Networked Predictive Control Method Using an Allowable Time Delay." Journal of Advanced Computational Intelligence and Intelligent Informatics 26, no. 5 (September 20, 2022): 768–75. http://dx.doi.org/10.20965/jaciii.2022.p0768.
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An event-triggered network predictive control method, which uses allowable time delays, was developed for networked control systems with random network delays, packet disorders, and packet dropouts in the feedback and forward channels. In this method, random communication constraints are uniformly treated as a time delay at each time instant. Subsequently, based on a time-delay state feedback control law, the proposed method is used to actively compensate for the time delay that exceeds the allowable. In addition, the introduction of an event-triggered mechanism reduces communication loads and saves network resources. A necessary and sufficient stability condition for the closed-loop system is provided, which is independent of random time delays and is related to the allowable delay. Finally, the simulation results of the two systems verified the effectiveness of the proposed method.
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HEMMINGER, THOMAS L., and CARLOS A. POMALAZA-RAEZ. "USING NEURAL NETWORKS TO SOLVE THE MULTICAST ROUTING PROBLEM IN PACKET RADIO NETWORKS." International Journal of Neural Systems 07, no. 05 (November 1996): 617–26. http://dx.doi.org/10.1142/s0129065796000609.
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The primary function of a packet radio network is the efficient transfer of information between source and destination nodes using minimal bandwidth and end-to-end delay. Many researchers have investigated the problem of minimizing the end-to-end delay from a single source to a single destination for a variety of networks; however, very little work is reported about routing mechanisms for the common case where a particular information packet is intended to be sent to more than one destination in the network. This is known as multicasting. A simplified version of the problem is to ignore the packet delay at each node, then the problem becomes one of finding solutions which require the least number of transmissions. Determination of an optimal solution is NP-complete meaning that suboptimal solutions are frequently tolerated. The problem becomes more rigorous if packet delays are included in the network topology. This paper describes a practical technique for the computation of optimum or near optimum solutions to the multicasting problem with and without packet delay. The method is based on the Hopfield neural network and experiment has shown this method to yield near optimal solutions while requiring a minimum of CPU time.
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Jarin, Asril, Suryadi Suryadi, and Kalamullah Ramli. "Packet Delay Distribution Model for Investigating Delay of Network Speech Recognition." Indonesian Journal of Electrical Engineering and Computer Science 5, no. 1 (January 1, 2017): 11. http://dx.doi.org/10.11591/ijeecs.v5.i1.pp11-18.
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Unlike multimedia streaming applications that require a smooth playback at the client, application of network speech recognition (NSR) that recognizes speech signal in a sentence-by-sentence manner might tolerate an acceptable delay. The acceptable delay is a user-defined time in which the entire sentence data should be received by the server. We proposed a calculation method to investigate the acceptable delay of network speech recognition that employs a speech segmenter to send speech signal sentence-by-sentence over TCP channel to the server. The calculation multiplies the mean packet delay of TCP flow at steady-state with the number of created packets. For validation we implemented a MATLAB program and solved it using 2500 Indonesian speech sentences. The results were then compared with the results of our previous model that used a transient analysis method. It was found that this calculation method is not appropriate due to the transient behavior of the streaming sentences.
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Said, Muhammad Ali, Setyorini Setyorini, and Erwid M. Jadied. "Analysis of IPSec Implementation on Dynamic Multipoint VPN Protocol Using Routing Border Gateway Protocol." Building of Informatics, Technology and Science (BITS) 4, no. 2 (September 22, 2022): 595–605. http://dx.doi.org/10.47065/bits.v4i2.1836.
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Dynamic Multipoint Virtual Private Network (DMVPN) technology is one of Cisco's solutions to overcome the limitations of VPN scalability. DMVPN has a combination of components: Next hop resolution protocol (NHRP), Multipoint Generic Routing Encapsulation (mGRE) and Routing protocol. This research implements a simple network consisting of Hub, Spoke1, Spoke2, Lan1, Lan2 and Lan3 using the GNS3 simulator. This study compares the performance of IPSec and without IPsec on DMVPN using the BGP Routing protocol on performance parameters namely delay, throughput, jitter and packet loss to evaluate the security impact of the DMVPN network. The results of this study indicate that IPSec DMVPN has an effect on sending UDP packets which have a throughput value without IPSec of 5082.18 kbit/s while IPSec's throughput is 5034.40 kbit/s. The value of packet loss without IPSec has a value of 7.54% while IPSec has a value of 4.79%. The results of the jitter value have the same value. The delay value without IPSec has a value of 0.183s while the IPSec delay value is 0.410s. TCP packet delivery has a throughput value without IPSec is 1139.16 kbit/s while the IPSec throughput value is 1105.20 kbit/s. The results of the packet loss value and the jitter value have the same value. The delay value without IPSec has a value of 0.185s while the IPSec delay value is 0.187s.
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Yang, Shuna, and Norvald Stol. "A novel delay line buffering architecture for asynchronous optical packet switched networks." International Journal of Information, Communication Technology and Applications 1, no. 1 (March 9, 2015): 69–82. http://dx.doi.org/10.17972/ajicta20151112.
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Optical buffering is one major challenge in realizing all-optical packet switching. In this paper we focus on a delay-line buffer architecture, named a Multiple-Input Single-Output (MISO) optical buffer, which is realized by cascaded fiber delay lines (FDLs). This architecture reduces the physical size of a buffer by up to an order of magnitude or more by allowing reuse of its delay line elements. We consider the MISO buffers in a network scenario where the incoming packets are asynchronous and of fixed length. A novel Markov model is developed to analyze the performance of our buffering scheme, in terms of packet loss ratio, average packet delay and the output link utilization. Both simulation and analytical results show that the length value of basic FDL element will significantly affect the performance of this buffer. This paper gives clear guidelines for designing optimal basic FDL lengths under different network scenarios. It is noticeable that this optimal length value is independent of the buffer sizes for specific traffic load and pattern.
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Ezema, Chukwuedozie N., Chinazam Ezemab C. Ezemab, and Chukwuebuka B. Umezinwac. "TRAFFIC DESIGN MODEL FOR OPTIMUM PERFORMANCE ENHANCEMENTS OF IEEE 802.11B QOS PARAMETERS." European Journal of Technology 1, no. 2 (March 31, 2017): 91. http://dx.doi.org/10.47672/ejt.228.
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Purpose: The objective of this research is to design a traffic model for optimum performance analysis and enhancements of IEEE 802.11b QOS parameters. It investigated the effect of traffic distribution on the quality of service parameter on WLAN, in other to establish enhanced WLAN Performance. In order to evaluate the mean packet delay and network throughput performance, this research has presented a traffic loading for MAC DCF which supports WLAN’s QOS Parameters.Methodology: Based on this model, a computer simulation model over a MATLAB Simulink was developed.Results: In this research, results show that the IEEE 802.11b does not perform well in terms of high throughput, and low mean delay at high traffic load conditions. Furthermore, it was also shown that the mean packet delay of arrived packets decreases as the number of workstations decreases, but at saturation, it was shown that throughput decreases, mean packet delay increases. Therefore, to achieve a better enhanced network performance, it was observed that IEEE 802.11b WLAN QOS parameters can be improved to a maximum throughput.
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Li, Fuliang, Xingwei Wang, Tian Pan, and Jiahai Yang. "A Case Study of IPv6 Network Performance: Packet Delay, Loss, and Reordering." Mathematical Problems in Engineering 2017 (2017): 1–10. http://dx.doi.org/10.1155/2017/3056475.
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Internet Protocol (IP) is used to identify and locate computers on the Internet. Currently, IPv4 still routes most Internet traffic. However, with the exhausting of IPv4 addresses, the transition to IPv6 is imminent, because, as the successor of IPv4, IPv6 can provide a larger available address space. Existing studies have addressed the notion that IPv6-centric next generation networks are widely deployed and applied. In order to gain a deep understanding of IPv6, this paper revisits several critical IPv6 performance metrics. Our extensive measurement shows that packet delay and loss rate of IPv6 are similar to IPv4 when the AS-level paths are roughly the same. Specifically, when the link utilization exceeds a threshold, for example, 0.83 in our study, variation of packet delay presents a similar pattern with the variation of link utilization. If packet delay of a path is large, packet-loss rate of that path is more likely to fluctuate. In addition, we conduct a first-ever analysis of packet reordering in IPv6 world. Few IPv6 probe packets are out-of-order and the reordering rate is 2.3⁎10-6, which is much lower than that of 0.79% in IPv4 world. Our analysis consolidates an experimental basis for operators and researchers of IPv6 networks.
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Peng, Mengyu, Wei Liu, Tian Wang, and Zhiwen Zeng. "Relay Selection Joint Consecutive Packet Routing Scheme to Improve Performance for Wake-Up Radio-Enabled WSNs." Wireless Communications and Mobile Computing 2020 (January 4, 2020): 1–32. http://dx.doi.org/10.1155/2020/7230565.
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Reducing energy consumption, increasing network throughput, and reducing delay are the pivot issues for wake-up radio- (WuR-) enabled wireless sensor networks (WSNs). In this paper, a relay selection joint consecutive packet routing (RS-CPR) scheme is proposed to reduce channel competition conflicts and energy consumption, increase network throughput, and then reduce end-to-end delay in data transmission for WuR-enabled WSNs. The main innovations of the RS-CPR scheme are as follows: (1) Relay selection: when selecting a relay node for routing, the sender will select the node with the highest evaluation weight from its forwarding node set (FNS). The weight of the node is weighted by the distance from the node to sink, the number of packets in the queue, and the residual energy of the node. (2) The node sends consecutive packets once it accesses the channel successfully, and it gives up the channel after sending all packets. Nodes that fail the competition sleep during the consecutive packet transmission of the winner to reduce collisions and energy consumption. (3) Every node sets two thresholds: the packet queue length threshold Nt and the packet maximum waiting time threshold Tt. When the corresponding value of the node is greater than the threshold, the node begins to contend for the channel. Besides, to make full use of energy and reduce delay, the threshold of nodes which are far from sink is small while that of nodes which are close to sink is large. In such a way, nodes in RS-CPR scheme will select those with much residual energy, a large number of packets, and a short distance from sink as relay nodes. As a result, the probability that a node with no packets to transmit becomes a relay is very small, and the probability that a node with many data packets in the queue becomes a relay is large. In this strategy, only a few nodes in routing need to contend for the channel to send packets, thereby reducing channel contention conflicts. Since the relay node has a large number of data packets, it can send many packets continuously after a successful competition. It also reduces the spending of channel competition and improves the network throughput. In summary, RS-CPR scheme combines the selection of relay nodes with consecutive packet routing strategy, which greatly improves the performance of the network. As is shown in our theoretical analysis and experimental results, compared with the receiver-initiated consecutive packet transmission WuR (RI-CPT-WuR) scheme and RI-WuR protocol, the RS-CPR scheme reduces end-to-end delay by 45.92% and 65.99%, respectively, and reduces channel collisions by 51.92% and 76.41%. Besides, it reduces energy consumption by 61.24% and 70.40%. At the same time, RS-CPR scheme improves network throughput by 47.37% and 75.02%.
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Sabbineni, Harshavardhan, and Krishnendu Chakrabarty. "An Energy-Efficient Data Delivery Scheme for Delay-Sensitive Traffic in Wireless Sensor Networks." International Journal of Distributed Sensor Networks 6, no. 1 (January 1, 2010): 792068. http://dx.doi.org/10.1155/2010/792068.
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We propose a novel data-delivery method for delay-sensitive traffic that significantly reduces the energy consumption in wireless sensor networks without reducing the number of packets that meet end-to-end real-time deadlines. The proposed method, referred to as SensiQoS, leverages the spatial and temporal correlation between the data generated by events in a sensor network and realizes energy savings through application-specific in-network aggregation of the data. SensiQoS maximizes energy savings by adaptively waiting for packets from upstream nodes to perform in-network processing without missing the real-time deadline for the data packets. SensiQoS is a distributed packet scheduling scheme, where nodes make localized decisions on when to schedule a packet for transmission to meet its end-to-end real-time deadline and to which neighbor they should forward the packet to save energy. We also present a localized algorithm for nodes to adapt to network traffic to maximize energy savings in the network. Simulation results show that SensiQoS improves the energy savings in sensor networks where events are sensed by multiple nodes, and spatial and/or temporal correlation exists among the data packets. Energy savings due to SensiQoS increase with increase in the density of the sensor nodes and the size of the sensed events.
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Jurca, Dan, and Pascal Frossard. "Packet Media Streaming with Imprecise Rate Estimation." Advances in Multimedia 2007 (2007): 1–8. http://dx.doi.org/10.1155/2007/39524.
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We address the problem of delay-constrained streaming of multimedia packets over dynamic bandwidth channels. Efficient streaming solutions generally rely on the knowledge of the channel bandwidth, in order to select the media packets to be transmitted, according to their sending time. However, the streaming server usually cannot have a perfect knowledge of the channel bandwidth, and important packets may be lost due to late arrival, if the scheduling is based on an over-estimated bandwidth. Robust media streaming techniques should take into account the mismatch between the values of the actual channel bandwidth and its estimation at the server. We address this rate prediction mismatch by media scheduling with a conservative delay, which provides a safety margin for the packet delivery, even in the presence of unpredicted bandwidth variations. We formulate an optimization problem whose goal is to obtain the optimal value for the conservative delay to be used in the scheduling process, given the network model and the actual playback delay imposed by the client. We eventually propose a simple alternative to the computation of the scheduling delay, which is effective in real-time streaming scenarios. Our streaming method proves to be robust against channel prediction errors, and performs better than other robustness mechanisms based on frame reordering strategies.
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G.RAJAVARMAN, G. RAJAVARMAN. "Analysis and Improve Packet Delay Problem in Wireless Sensor Networks." Indian Journal of Applied Research 4, no. 4 (October 1, 2011): 210–13. http://dx.doi.org/10.15373/2249555x/apr2014/64.
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Bharti, Rajendra Kumar, V. Bhoopathy, Parul Bhanarkar, Kanahaiya Lal Ambashtha, K. Priya, C. Anand Deva Durai, Manam Vamsi Krishna, P. Joel Josephson, and Kibebe Sahile. "Routing Path Selection and Data Transmission in Industry-Based Mobile Communications Using Optimization Technique." Wireless Communications and Mobile Computing 2022 (July 21, 2022): 1–9. http://dx.doi.org/10.1155/2022/5431413.
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In a mobile network, nodes are share data packets; sometimes, that packets are totally flooding. The packet dropping node does not easily detect for routing time instance. The node trust level is minimum causing the packet loss; it affects the entire network performance, and it reduces throughput and increases communication overhead. Proposed exhaustive routing path allocation (ERP) technique is applied to select the legitimate node for broadcasting the data packets completely. The attacker nodes of that flooding packets are detected by using the legitimate detector which are present in network environment. The node credence level evaluation algorithm is planned to estimating each and every node authority range, whether the nodes have higher credence level basis efficient packet transmission in wireless nodes; otherwise, nodes have lesser credence level basis in effective packet broadcasting. These higher credence level nodes are assigned for communication process in movable network. It improves the throughput and minimizes the communication overhead. The performance metrics of the parameters are delay, communication overhead, throughput, network lifetime, energy consumption, and packet loss.
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Marais, Jaco, Reza Malekian, Ning Ye, and Ruchuan Wang. "A Review of the Topologies Used in Smart Water Meter Networks: A Wireless Sensor Network Application." Journal of Sensors 2016 (2016): 1–12. http://dx.doi.org/10.1155/2016/9857568.
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This paper presents several proposed and existing smart utility meter systems as well as their communication networks to identify the challenges of creating scalable smart water meter networks. Network simulations are performed on 3 network topologies (star, tree, and mesh) to determine their suitability for smart water meter networks. The simulations found that once a number of nodes threshold is exceeded the network’s delay increases dramatically regardless of implemented topology. This threshold is at a relatively low number of nodes (50) and the use of network topologies such as tree or mesh helps alleviate this problem and results in lower network delays. Further simulations found that the successful transmission of application layer packets in a 70-end node tree network can be improved by 212% when end nodes only transmit data to their nearest router node. The relationship between packet success rate and different packet sizes was also investigated and reducing the packet size with a factor of 16 resulted in either 156% or 300% increases in the amount of successfully received packets depending on the network setup.
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Shelkovoy, D. V., and A. A. Chernikov. "Simulation modeling of packet switching network segment functioning." Issues of radio electronics, no. 12 (December 28, 2019): 75–82. http://dx.doi.org/10.21778/2218-5453-2019-12-75-82.
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The testing results of required channel resource mathematical estimating models for the for serving the proposed multimedia load in packet-switched communication networks are presented in the article. The assessment of the attainable level of quality of service at the level of data packet transportation was carried out by means of simulation modeling of the functioning of a switching node of a communication network. The developed modeling algorithm differs from the existing ones by taking into account the introduced delay for processing each data stream packet arriving at the switching node, depending on the size of the reserved buffer and the channel resource for its maintenance. A joint examination of the probability of packet loss and the introduced delay in the processing of data packets in the border router allows a comprehensive assessment of the quality of service «end to end», which in turn allows you to get more accurate values of the effective data transmitted rate by aggregating flows at the entrance to the transport network.
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Lozhkovskyi, A. G., and K. D. Guliaiev. "CALCULATING METHOD OF THE PACKET NETWORK ACCESS TRAFFIC CAPACITY FOR IoT DEVICES." Proceedings of the O.S. Popov ОNAT 1, no. 2 (December 31, 2020): 31–40. http://dx.doi.org/10.33243/2518-7139-2020-1-2-31-40.
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When creating packet networks of a new generation, the problem of calculating the throughput of broadband multiservice access networks arises. In practice, for this they use mathematical modeling, or without proper justification, the traditional formulas of the theory of information distribution. The well-known analytical solution for ATM technology is extremely cumbersome and practically not used. Today, there is not generally accepted analytical or engineering method for solving the problem. At the same time, telecommunication networks must develop to provide all the necessary conditions for the practical application of the concept of the Internet of Things (IoT). One of these conditions is the maintenance of multiservice traffic with specified quality of service indicators. The paper developed a method for calculating the bandwidth or the number of conditional channels of a packet access network for IoT devices. In this case, the calculation of the bandwidth of the access network of IoT devices is performed at the level of calls and packets separately. At the level of calls from IoT devices, the Engset model is used for traffic due to the small number of groups of the devices themselves, and at the packet level, the model of self-similar flow is applied. Calculation of quality of service characteristics in a packet communication network is reduced to determining the Hurst coefficient of self-similarity of traffic, after which the average number of packets in the system is calculated using the well-known Norros formula. Other characteristics, such as the average number of packets in the queue, the average residence time of packets in the system and the average delay time of packets in a single-channel system, are calculated based on their functional relationship with the previously calculated average number of packets in the system. Based on the approximation of the distribution function of the system states, the probability of waiting for packet servicing and the average delay time of packets in the packet switch queue are calculated.
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Cui, Laizhong, Nan Lu, and Fu Chen. "Exploring a QoS Driven Scheduling Approach for Peer-to-Peer Live Streaming Systems with Network Coding." Scientific World Journal 2014 (2014): 1–10. http://dx.doi.org/10.1155/2014/513861.
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Most large-scale peer-to-peer (P2P) live streaming systems use mesh to organize peers and leverage pull scheduling to transmit packets for providing robustness in dynamic environment. The pull scheduling brings large packet delay. Network coding makes the push scheduling feasible in mesh P2P live streaming and improves the efficiency. However, it may also introduce some extra delays and coding computational overhead. To improve the packet delay, streaming quality, and coding overhead, in this paper are as follows. we propose a QoS driven push scheduling approach. The main contributions of this paper are: (i) We introduce a new network coding method to increase the content diversity and reduce the complexity of scheduling; (ii) we formulate the push scheduling as an optimization problem and transform it to a min-cost flow problem for solving it in polynomial time; (iii) we propose a push scheduling algorithm to reduce the coding overhead and do extensive experiments to validate the effectiveness of our approach. Compared with previous approaches, the simulation results demonstrate thatpacket delay,continuity index,andcoding ratioof our system can be significantly improved, especially in dynamic environments.
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Bocharova, Irina, Boris Kudryashov, Nikita Lyamin, Erik Frick, Maben Rabi, and Alexey Vinel. "Low Delay Inter-Packet Coding in Vehicular Networks." Future Internet 11, no. 10 (October 11, 2019): 212. http://dx.doi.org/10.3390/fi11100212.
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In Cooperative Intelligent Transportation Systems (C-ITSs), vehicles need to wirelessly connect with Roadside units (RSUs) over limited durations when such point-to-point connections are possible. One example of such communications is the downloading of maps to the C-ITS vehicles. Another example occurs in the testing of C-ITS vehicles, where the tested vehicles upload trajectory records to the roadside units. Because of real-time requirements, and limited bandwidths, data are sent as User Datagram Protocol (UDP) packets. We propose an inter-packet error control coding scheme that improves the recovery of data when some of these packets are lost; we argue that the coding scheme has to be one of convolutional coding. We measure performance through the session averaged probability of successfully delivering groups of packets. We analyze two classes of convolution codes and propose a low-complexity decoding procedure suitable for network applications. We conclude that Reed–Solomon convolutional codes perform better than Wyner–Ash codes at the cost of higher complexity. We show this by simulation on the memoryless binary erasure channel (BEC) and channels with memory, and through simulations of the IEEE 802.11p DSRC/ITS-G5 network at the C-ITS test track AstaZero.
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Lebl, Aleksandar, Dragan Mitic, Vladimir Matic, and Zarko Markov. "Model for speech signal quality estimation in packet network of electricity supply industry." Serbian Journal of Electrical Engineering 16, no. 3 (2019): 405–17. http://dx.doi.org/10.2298/sjee1903405l.
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In this paper we present the dilemma dealing with the choice of speech segment duration and the calculation of speech signal quality in the packet telephone network of Electricity supply industry (ESI). The analyzed signal is packetized by one packet. The characteristic of this network is that disturbances of long duration in power system produce burst packet loss. The longer speech segments increase the packet delay, but decrease the number of lost packets in one burst, and vice versa. Here we present why it is better to choose speech segments of short duration. Also, we suggest the corrected method for the calculation of speech signal quality in packet network under the influence of long duration disturbances.
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Buranova, M. "Assessing Jitter when Processing Traffic in the G/G/1 System." Telecom IT 8, no. 2 (June 2020): 12–19. http://dx.doi.org/10.31854/2307-1303-2020-8-2-11-19.
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Analysis of the parameters of the functioning of IP-networks in the processing of multimedia streams is a very important task. There are many approaches to assessing the quality of service parameters in the G/G/1 system. Changing the delay of packets on the network is a very important parameter that deter-mines the quality of traffic processing. The change in delay is usually defined either as packet jitter or as a variation in packet delay. At the same time, the required accuracy in determining the parameters can be obtained quite accurately, since all decisions are mainly based on certain assumptions. The paper presents an approach to determining the change in packet delay in the G/G/1 system as packet jitter. The basis of the presented approach is the approximation of arbitrary distributions by hyperexponential distributions, i.e. modeling of the G/G/1 system by the H2/H2/1 system. The solution to the jitter estima-tion problem is to determine the distribution parameters. To evaluate the parameters of hyperexponen-tial distributions, the EM algorithm is used. As the traffic studied, a multimedia stream registered on a real network was used. An analytical estimate of jitter in the G/G/1 system is obtained. The results ob-tained are applicable for independent flows and for random variables uncorrelated in the structure of each sequence.
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Nadia Wan Abdullah, Wan Aida, Naimah Yaakob, R. Badlishah Ahmad, Mohamed Elshaikh Elobaid, and Siti Asilah Yah. "Corrupted packets discarding mechanism to alleviate congestion in wireless body area network." Indonesian Journal of Electrical Engineering and Computer Science 14, no. 2 (May 1, 2019): 581. http://dx.doi.org/10.11591/ijeecs.v14.i2.pp581-587.
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<span>Generation of high traffic from continuous sensing and collection of medical data from various biosensors on multiple body is most likely to occur in the Wireless Body Area Network (WBAN). This could be a factor to the congestion in the network. Occurrence of congestion would collapse the performances in the WBAN network in terms of increment in delay, high packets loss, reduction in throughput and packet deliver ratio (PDR). The crucial concerns in WBAN are prevention from the loss of critical data and longer delay in the network as they could result to late delivery of medical treatment and possibility of the increase in mortality. Therefore, this study proposes a mechanism to alleviate the congestion from happening in the first place through discarding the corrupted packets before the beginning of data transmission to the base station. Extensive simulations are done in OMNeT+ to analyze the performance of the proposed mechanism by varying traffic from low to high under different number of nodes and constant Bit Error Rate (BER) and packet size. From the finding, it can be concluded that the proposed mechanism shows better performances in terms of low delay and packet loss as well as high throughput and PDR compared to typical WBAN.</span>
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Wu, Li Zhen, and Xiao Hong Hao. "A Gain-Schedule Adaptive LQG Control for Networked Systems with Time Delay and Packet Dropout." Applied Mechanics and Materials 236-237 (November 2012): 1067–71. http://dx.doi.org/10.4028/www.scientific.net/amm.236-237.1067.
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This paper studies the coordinate optimization control problem of networked control systems (NCSs) with random time delay and packet dropout. A discrete-time system model of NCSs with time-delays and data packet dropout is proposed. A method of state estimation base on extra kalman filter is given. Then a gain-schedule adaptive LQG control strategy base on effective delay-estimation online is proposed. The result illustrate that the effectiveness of the proposed controller design and the satisfactory performance of the system.
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Han, Shi Qian, Jin Na Li, and Mao Quan Wu. "Novel Modeling Method for Networked Control Systems with Packet Disordering." Advanced Materials Research 433-440 (January 2012): 2491–97. http://dx.doi.org/10.4028/www.scientific.net/amr.433-440.2491.
Full textAbstract:
This paper focuses on the modeling problem for networked control systems subject to packet disordering. Both network-induced delay and packet loss are taken into consideration. In constructing the model, we first present the novel model for single input and single output (SISO) networked control systems based on the displacement values of packets in terms of switched systems theory, and then which can be extended to the case in multiple input and multiple output (MIMO) networked control systems with packet disordering. The merits or advantages of the modeling method proposed include describing fully the phenomenon of packet disordering, guaranteeing the newest signals to be executed and extensive applicability.