A QoS Based Formal Model for Software Defined Network

2020 ◽  
Vol 10 (3) ◽  
pp. 395-401
Author(s):  
Vivek Srivastava ◽  
Ravi Shankar Pandey
Keyword(s):  
Real Time ◽  
Channel Capacity ◽  
Packet Loss ◽  
Formal Model ◽  
Optimal Path ◽  
Network Size ◽  
Central Controller ◽  
High Traffic Volume

Background & Objective: Software-Defined Networks (SDN) decouple the responsibility of data plane, control plane and aggregates responsibilities at the controller. The controller manages all the requests generated from distributed switches to get the optimal path for sending data from source to destination using load balancing algorithms. The guarantee of packet reachability is a major challenge in real time scenario of a SDN which depends on components of network infrastructure as switches, a central controller, channel capacity and server load. The success of this aggregation and packet reachability demand is a high Quality of Service (QoS) requirement in terms of throughput, delay and packet loss due to high traffic volume and network size. This QoS has two perspectives one is required other is a computation of real QoS value. Methods: In this paper, we have presented the QoS based formal model of SDN to compute and to investigate the role of the real QoS value. This formal model includes QoS on the basis of packet movement hop by hop which is a real-time QoS. The hop by hop packet movement reliability has been computed using channel capacity and server load which is an abstraction of throughput, delay, and packet loss. The effect of channel capacity and server load can be varying using different values of the weight factor. We have also considered an equal role of channel capacity and server load to compute reliability. This QoS helps to the controller to match with required QoS to decide the better path. Conclusion: Our results finds the reliable path based on channel capacity and server load of the network. Also, results showed that the reliability of the network and controller which are based on the reliability of the packet delivery between two nodes.

An Improved Algorithm of Adaptive Multi-Pulse AMR-WB PLC

2013 ◽  
Vol 321-324 ◽  
pp. 2754-2759
Author(s):  
Jun Ying Jia ◽  
Hu Lin ◽  
Jian Wei Sun
Keyword(s):  
Real Time ◽  
Packet Loss ◽  
Mobile Terminal ◽  
Decoding Algorithm ◽  
Basic Principles ◽  
Voice Communication ◽  
Speed Up ◽  
Small Bandwidth ◽  
Adaptive Codebook

Packet loss is common in internet, when use IMS mobile terminal accessed by wireless to do real-time voice communication, the quality of speech is affected a lot. This text summarizes the basic principles of AMR-WB encoding and decoding algorithm and main technology of PLC. Based on AMR-WB encoding and decoding algorithm, it improves an algorithm of multi-plus PLC. The algorithm described in this text can speed up the recovery of adaptive-codebook by small bandwidth cost and effectively prevent the spread of errors, improve the quality of speech.

scholarly journals Analysis Tunneling IPv4 and IPv6 on VoIP Network

2018 ◽  
pp. 337-344 ◽  
Author(s):  
Muhammad Ismu Haji ◽  
Sugeng Purwantoro E.S.G.S ◽  
Satria Perdana Arifin
Keyword(s):  
Real Time ◽  
Packet Loss ◽  
Test Results ◽  
The Real ◽  
Ip Addresses ◽  
Internet Users ◽  
Jitter Delay ◽  
Ipv4 Address

Using of IP addresses is currently still using IPv4. Meanwhile, the availability of the IPv4 address is gradually diminishes. IPv4 has a limited address capacity. IPv6 was developed with a capacity greater than IPv4. Connect between IPv4 and IPv6 without having to interfere with the existing infrastructure. So, methods like tunneling are needed. Tunneling builds a way that IPv4 and IPv6 can communicate. 6to4 tuning makes IPv6 able to communicate with IPv4 over IPv4 infrastructure. Real time communication is needed by internet users to be able to connect to each other. One of the real time communications is VoIP. To find out the quality of tunneling implemented on a VoIP network, it will analyze QoS such as delay, packet loss, and jitter. Delay obtained is 20,01ms for IPv4, 19,99ms for IPv6 and 20,03ms for 6to4. Packet loss obtained 0,01% for IPv4, IPv6 0,01% and 6to4 0,08%. The obtained jitter is 7,96ms for IPv4, IPv6 7.39ms, and 8,48 for 6to4. The test results show that using IPv6 gets a better QoS value than using IPv4 and 6to4 tunneling. The results using 6to4 tunneling obtained the highest QoS value between IPv4 and IPv6. Implementation using 6to4 tunneling results in high results because, IPv6 packets that are sent are wrapped into the IPv4 form to get through the IPv4 infrastructure. 

scholarly journals Interconnecting Haptic Interfaces with High Update Rates through the Internet

2018 ◽  
Vol 1 (4) ◽  
pp. 51
Author(s):  
George Kokkonis ◽  
Kostas Psannis ◽  
Sotirios Kontogiannis ◽  
Petros Nicopolitidis ◽  
Manos Roumeliotis ◽  
...  
Keyword(s):  
Quality Of Service ◽  
Performance Evaluation ◽  
Real Time ◽  
Packet Loss ◽  
The Internet ◽  
Haptic Interfaces ◽  
Transport Protocols ◽  
Delay Jitter ◽  
Packet Arrival

Real-time transferring of the haptic sense over the Internet is quite a challenging task. This paper outlines the proposed protocols for transferring haptic streams over the Internet. Moreover, it describes the Quality of Service requirements that a network has to fulfill in order to successfully use haptic interfaces with high update rates over the Internet. Extensive simulations and experiments for the performance evaluation of transport protocols for real-time transferring haptic data are carried out. Complements between simulation and real world experiments are discussed. The metrics that are measured for the performance evaluation are delay, jitter, throughput, efficiency, packet loss and one proposed by the authors, packet arrival deviation. The simulation tests reveal which protocols could be used for the transfer of real-time haptic data over the Internet.

scholarly journals Contiki-Based IEEE 802.15.4 Channel Capacity Estimation and Suitability of Its CSMA-CA MAC Layer Protocol for Real-Time Multimedia Applications

2015 ◽  
Vol 2015 ◽  
pp. 1-9 ◽  
Author(s):  
Muhammad Omer Farooq ◽  
Thomas Kunz
Keyword(s):  
Real Time ◽  
Channel Capacity ◽  
Packet Loss ◽  
Ieee 802.15.4 ◽  
Mac Protocol ◽  
Multimedia Applications ◽  
Mac Layer ◽  
Medium Access ◽  
Available Bandwidth ◽  
Bandwidth Guarantees

Real-time multimedia applications require quality of service (QoS) provisioning in terms of bounds on delay and packet loss along with soft bandwidth guarantees. The shared nature of the wireless communication medium results in interference. Interference combined with the overheads, associated with a medium access control (MAC) protocol, and the implementation of a networking protocol stack limit the available bandwidth in IEEE 802.15.4-based networks and can result in congestion, even if the transmission rates of nodes are well below the maximum bandwidth supported by an underlying communication technology. Congestion degrades the performance of admitted real-time multimedia flow(s). Therefore, in this paper, we experimentally derive the IEEE 802.15.4 channel capacity using an unslotted CSMA-CA MAC protocol. We experimentally derive channel capacity for two cases, that is, when the CSMA-CA protocol is working without ACKs and when it is working with ACKs. Moreover, for both cases, we plot the relationship of offered data load with delay and packet loss rate. Simulation results demonstrate that the parameters that affect the choice of a CSMA-CA MAC layer protocol are end-to-end delay and packet loss requirements of a real-time multimedia flow, data load within the interference range of transmitters along the forwarding path, and length of the forwarding path.

scholarly journals Dynamic real-time capacity constrained routing algorithm for evacuation planning problem

2020 ◽  
Vol 20 (3) ◽  
pp. 1388
Author(s):  
Jawad Abusalama ◽  
Sazalinsyah Razali ◽  
Yun-Huoy Choo ◽  
Lina Momani ◽  
Abdelrahman Alkharabsheh
Keyword(s):  
Real Time ◽  
Transportation Network ◽  
Optimal Path ◽  
Computational Cost ◽  
Optimal Solution ◽  
Emergency Evacuation ◽  
Network Size ◽  
Evacuation Planning ◽  
Planning Problem ◽  
Capacity Constrained

<span>Usually, disasters occurred over a relatively short time in anytime and anywhere. Most occupancies haven’t absolute knowledge about the prevention or safety consciousness to deal with disasters. During disaster occurred, evacuation processes are conducted to save people life, and if there is no appropriate evacuation plan, the situation will become worse. Thus, finding optimal planning to evacuate the occupancy people is critical in many cases i.e. emergency evacuation. In this paper, a Dynamic Real-Time Capacity Constrained Routing (DRTCCR) Algorithm has been proposed and analyzed. Such algorithm will investigate the capacity constraints of the evacuation network in real-time by modelling the capacities at the time of series to improve current solutions of the evacuation planning problem.  Such algorithm will produce an optimal solution for evacuation planning problem. Performance evaluation on many network models illustrates that the DRTCCR algorithm improves the previous evacuation planning by reducing the evacuation time as well as the computational cost. In addition, DRTCCR algorithm has the ability to recalculate and find out the optimal path dynamically in real-time irrespective the number of trapped people as well as the transportation network size. Analytical experiments have been done and illustrate the efficiency of the proposed algorithm.</span>

An Adaptive Security Framework for the Internet of Things Applications Based on the Contextual Information

2019 ◽  
pp. 244-267
Author(s):  
Harsuminder Kaur Gill ◽  
Anil Kumar Verma ◽  
Rajinder Sandhu
Keyword(s):  
Internet Of Things ◽  
Real Time ◽  
Security Policy ◽  
Learning Algorithm ◽  
Contextual Information ◽  
Service Access ◽  
Current Context ◽  
User Demand

With the growth of Internet of Things and user demand for personalized applications, context-aware applications are gaining popularity in current IT cyberspace. Personalized content, which can be a notification, recommendation, etc., are generated based on the contextual information such as location, temperature, and nearby objects. Furthermore, contextual information can also play an important role in security management of user or device in real time. When the context of a user or device changes, the security mechanisms should also be updated in real time for better performance and quality of service. Access to a specific resource may also be dependent upon user's/device's current context. In this chapter, the role of contextual information for IoT application security is discussed and a framework is provided which auto-updates security policy of the device based on its current context. Proposed framework makes use of machine learning algorithm to update the security policies based on the current context of the IoT device(s).

scholarly journals Implementasi TVoIP Menggunakan Komunikasi WLAN Untuk Smartphone Android

2018 ◽  
Vol 25 (2) ◽  
Author(s):  
Indrasto Jati P ◽  
S.El Yumin
Keyword(s):  
Real Time ◽  
Packet Loss ◽  
Wireless Lan ◽  
Access Point ◽  
Quality Of Services ◽  
Streaming Server ◽  
Delay Jitter ◽  
Interval Delay

TV over IP merupakan salah satu aplikasi komunikasi multimedia yang memanfaatkan prosesstreaming dalam pengiriman paket-paket data videonya melalui jaringan Internet Protokol (IP). Karenaditerapkan pada jaringan yang berbasis IP, maka akan menggunakan transmisi secara real time yang dapatdibroadcast melalui wireless LAN. Smartphone android akan memberikan manfaat yang lebih karena sudahdilengkapi dengan perangkat wireless. Dalam makalah ini dibahas perancangan server TV over IP denganmenggunakan USB TV tuner untuk menangkap siaran televisi. Untuk membroadcast siaran televisi digunakanperangkat access point melalui jaringan wireless LAN. Pengguna smartphone android yang mempunyaiperangkat wirelss dapat mengakses siaran televisi yang dibroadcast oleh server. Dari implementasi yang telahdilakukan akan dianalisa kualitas layanan streaming atau Quality of Services (QoS) berupa throughput, delay,jitter, dan packet loss. Analisis perencanaan penambahan user dengan simulasi penambahan background trafficdan perencanaan jarak yang semakin jauh dari server akan menurunkan kualitas layanan. Nilai througput akanberbanding terbalik dengan packet loss, tetapi pada pengujian yang dilakukan nilai jitter tidak stabil karenaberpengaruh dari interval delay yang tidak teratur pada paket yang diterima. Jadi dapat disimpulkan bahwasemakin banyak user yang mengakses streaming server maka nilai kualitas layanan akan semakin menurunkarena bandwidth akan semakin kecil yang disebabkan oleh padatnya traffic pada wireless LAN. Denganperancangan TV over IP ini dapat mempermudah bagi pengguna perangkat yang mempunyai wireless sepertismartphone android untuk meengakses siaran televisi lokal di dalam area jangkauan wireless. Dengan layananTV yang berbasis IP akan menghasilkan gambar yang lebih interaktif. Karena merupakan layanan streamingmaka TV over IP rentan dengan kebutuhan bandwidth dengan jumlah kenaikan user dan juga padaperancangan wireless terbatas pada jarak tertentu.

scholarly journals Application Layer ARQ Algorithm for Real-Time Multi-Source Data Streaming in UAV Networks

Sensors ◽  
2021 ◽  
Vol 21 (17) ◽  
pp. 5763
Author(s):  
Mohammed Amin Lamri ◽  
Albert Abilov ◽  
Danil Vasiliev ◽  
Irina Kaisina ◽  
Anatoli Nistyuk
Keyword(s):  
Real Time ◽  
Packet Loss ◽  
Data Flow ◽  
Critical Role ◽  
Buffer Management ◽  
Base Station ◽  
Video Data ◽  
Application Layer ◽  
Delay Jitter

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.

scholarly journals ANALISIS PERFORMASI TRANSMISI VIDEO PADA JARINGAN MULTICAST DAN UNICAST DENGAN MENGGUNAKAN WLAN IEEE 802.11N

2020 ◽  
Vol 5 (2) ◽  
Author(s):  
Mutia Muliana ◽  
Rizal Munadi ◽  
Teuku Yuliar Arif
Keyword(s):  
Quality Of Service ◽  
Real Time ◽  
Video Streaming ◽  
Packet Loss ◽  
Bit Rate ◽  
User Datagram Protocol ◽  
Ieee 802.11N ◽  
Streaming Multicast

Abstrak— Perkembangan pemakaian internet semakin meluas khususnya dalam bidang jaringan. Saat ini orang berkomunikasi tidak hanya dengan suara maupun teks, tetapi juga secara visual maupu   n menggunakan video streaming. Penggunakan akses wireless sangat tinggi namun wilayah coverage wireless yang tersedia dalam satu gedung terbatas sehingga daya tangkap sinyal wireless berbeda  satu dengan yang lain. Penelitian ini bertujuan untuk menganalisis performasi multicast dan unicast  dengan melihat pengaruh bit rate dan terhadap kualitas layanan streaming menggunakan protocol Real Time Protocol (RTP) dan User Datagram Protocol (UDP). Pada peneletian menggunakan dua format video yang berbeda yaitu MPEG 4 dan H.264 dengan WLAN 802.11n menggunakan metode eksperimental untuk evaluasi kinerja dua protokol yang berbeda  dengan  parameter Quality of Service (Qos). Multicast bekerja dengan mengirim data  kepada  banyak  titik sekaligus dan unicast bekerja dengan mengirim data kepada satu client. Standar IEEE 802.11n digunakan untuk menguji performa wifi dalam mentransmisikan video streaming. Hasil dari penelitian menunjukkan dengan multicast menggunakan sebanyak 4 client yaitu mempertimbangkan QoS terbaik yaitu throughput tertinggi, delay terkecil dan packet loss minimum diusulkan penggunaan protokol RTP dengan format video MPEG 4 lebih baik pada sistem transmisi streaming secara Unicast.  Kata Kunci – Video Streaming, Multicast, Unicast,802.11n, QoS.

scholarly journals The moderating role of social network size in the temporal association between formal social participation and mental health: a longitudinal analysis using two consecutive waves of the Survey of Health, Ageing and Retirement in Europe (SHARE)

2020 ◽  
Author(s):  
Ziggi Ivan Santini ◽  
Paul E. Jose ◽  
Ai Koyanagi ◽  
Charlotte Meilstrup ◽  
Line Nielsen ◽  
...  
Keyword(s):  
Mental Health ◽  
Social Network ◽  
Social Participation ◽  
Social Ties ◽  
Network Size ◽  
Depression Symptoms ◽  
Social Network Size ◽  
Moderating Role

Abstract Introduction Previous studies have shown that engaging in formal social participation may protect against declining mental health, but social network size (the number of close social ties a person has) may moderate the relationship. We assessed the potential moderating role of social network size using longitudinal data. Methods Nationally representative data from two consecutive waves (2011, 2013) of the SHARE survey were analyzed. The data consisted of 38,300 adults from 13 European countries aged 50 years and older in 2011. Measures pertaining to formal social participation, social network size, quality of life, and depression symptoms were used. Multivariable linear regression models were conducted. Results The majority of participants (over 70% of the sample) had a social network size of four or less close social ties. We identified significant moderations in both models. Individuals with relatively few close social ties may have benefitted from formal social participation both in terms of reductions in depression symptoms and increases in quality of life, while formal social participation among those with many social ties did not appear to be beneficial, and may even to some extent have been detrimental. Conclusions Declines in mental health specifically among those with relatively few close social ties could potentially be prevented through the promotion of formal social participation. It is possible that such strategies could have a greater impact by specifically targeting individuals that are otherwise socially isolated. High levels of formal participation among those with relatively many close social ties may not be pragmatically beneficial.

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