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 Reconstruction of Generative Adversarial Networks in Cross Modal Image Generation with Canonical Polyadic Decomposition

2021 ◽  
Vol 2021 ◽  
pp. 1-9
Author(s):  
Ruixin Ma ◽  
Junying Lou ◽  
Peng Li ◽  
Jing Gao
Keyword(s):  
Mobile Terminal ◽  
Experimental Results ◽  
Generative Adversarial Networks ◽  
Image Generation ◽  
Adversarial Networks ◽  
Speed Up ◽  
Canonical Polyadic Decomposition

Generating pictures from text is an interesting, classic, and challenging task. Benefited from the development of generative adversarial networks (GAN), the generation quality of this task has been greatly improved. Many excellent cross modal GAN models have been put forward. These models add extensive layers and constraints to get impressive generation pictures. However, complexity and computation of existing cross modal GANs are too high to be deployed in mobile terminal. To solve this problem, this paper designs a compact cross modal GAN based on canonical polyadic decomposition. We replace an original convolution layer with three small convolution layers and use an autoencoder to stabilize and speed up training. The experimental results show that our model achieves 20% times of compression in both parameters and FLOPs without loss of quality on generated images.

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.

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.

Research on the Roller Intelligent Detection System Based on LabVIEW

2013 ◽  
Vol 438-439 ◽  
pp. 1084-1088
Author(s):  
Ummin Okumura ◽  
Yu Jie Qi ◽  
Yun Long ◽  
Tian Hang Zhang
Keyword(s):  
Nondestructive Testing ◽  
Real Time ◽  
Detection System ◽  
Detection Methods ◽  
Intelligent Detection ◽  
Speed Up ◽  
Compaction Control ◽  
And Control ◽  
Compaction Degree

Based on the platform of LabVIEW, a set of roller intelligent detecting system is developed. With this system, it is easy to realize functions of fast nondestructive testing of subgrade compaction degree, roller speed, rollers compaction trajectory, compaction times, GPS real-time positioning as well as saving and printing report forms. Compared with traditional detection methods, this detecting system can test and control on-site compaction quality much more easily, in order to speed up the construction progress, improve the quality of subgrade compaction, control and manage compaction work better.

scholarly journals A Statistical Approach to Adaptive Playout Scheduling in Voice Over Internet Protocol Communication

2018 ◽  
Vol 8 (5) ◽  
pp. 2926
Author(s):  
Priya Chandran ◽  
Chelpa Lingam
Keyword(s):  
Packet Loss ◽  
Loss Rate ◽  
Voice Quality ◽  
Internet Protocol ◽  
Packet Loss Rate ◽  
Voice Over Internet Protocol ◽  
Network Delay ◽  
Voice Communication ◽  
Voice Packets

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.

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.

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