Challenges in Quality of Service for Tomorrow's Networks

Author(s):  
Luiz A. DaSilva

The original communication networks were designed to carry traffic with homogeneous performance requirements. The telephone network carried real-time voice, with stringent latency bounds, and therefore used circuit-switched technologies with fixed bandwidth allocated to each call. Original data networks were used for electronic mail and file exchange, and therefore employed packet switching and provided best-effort service.

2002 ◽  
pp. 106-122
Author(s):  
Luiz A. DaSilva

Today’s networks support applications that deliver text, audio, images and video, often in real time and with a high degree of interactivity, using a common infrastructure. More often than not, traffic is carried over packet-switched networks that treat all data the same, under what is known as best-effort service. Packet switching can achieve very high efficiency through statistical multiplexing of data from numerous sources; however, due to the very nature of packet switching, one should expect fluctuations in throughput, delay, reliability, etc., for any given flow. The greater the statistical multiplexing capabilities, the greater the efficiency and also the greater the variability of achieved performance; in this sense, best-effort service provides maximum efficiency with highly unpredictable service quality. Clearly, not all traffic flows are created equal. Interactive web-based applications tend to be very sensitive to throughput, while real-time voice and video are sensitive to delay and jitter, and traditional data applications such as e-mail and file transfers are fairly insensitive to fluctuations in performance. The concept of quality of service (QoS) has evolved from the realization that in networks that carry heterogeneous traffic it makes sense to treat specific classes of traffic according to their specific needs.


Author(s):  
D. V. Shelkovoy ◽  
A. A. Chernikov

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.


IEEE Access ◽  
2021 ◽  
pp. 1-1
Author(s):  
Hafsa Bibi ◽  
Farrukh Zeeshan Khan ◽  
Muneer Ahmad ◽  
Anum Naseem ◽  
Tomasz Holynski ◽  
...  

2017 ◽  
Vol 3 (2) ◽  
pp. 249-254
Author(s):  
Darmawan Darmawan ◽  
Yayan Syafriyatno

Voice over IP (VoIP) adalah solusi komunikasi suara yang murah karena menggunakan jaringan IP dibanding penggunaan telephone analog yang banyak memakan biaya. Dalam penerapannya, VoIP mengalami permasalahan karena menggunakan teknologi packet switching yang mana penggunaannya bersamaan dengan paket data sehingga timbul delay, jitter, dan packet loss.  Pada penelitian ini, algoritma Low Latency Queuing (LLQ) diterapkan pada router cisco. Algoritma LLQ merupakan gabungan dari algoritma Priority Queuing (PQ) dan Class Based Weight Fair Queuing (CBWFQ) sehingga dapat memprioritaskan paket suara disamping paket data. Algoritma LLQ ini diujikan menggunakan codec GSM FR, G722, dan G711 A-law. Hasil pengujian didapatkan nilai parameter yang tidak jauh berbeda dan memenuhi standar ITU-T.G1010. Nilai delay rata - rata terendah yaitu ketika menggunakan codec G722 sebesar 20,019 ms tetapi G722 memiliki rata - rata jitter yang terbesar yaitu 0,986 ms.  Codec dengan jitter rata – rata terkecil adalah G711 A-law sebesar 0,838 ms. Packet loss untuk semua codec yang diujikan adalah 0%.  Throughput pada paket data terbesar saat menggunakan codec GSM FR yaitu 18,139 kbps. Codec yang direkomendasikan adalah G711 A-law karena lebih stabil dari segi jitter dan codec GSM FR cocok diimplementasikan pada jaringan yang memiliki bandwitdh kecil.


2018 ◽  
Vol 7 (2.28) ◽  
pp. 181
Author(s):  
Ali M. Al-Saegh

Building scheduling algorithms in satellite communication links became a necessity according to the typical problems that satellite networks suffers from, such as congestions, jamming, mobility, atmospheric impairment, and achieving the quality of service (QoS) requirements. However, building efficient algorithms needs several considerations that should be taken into account. Such as satellite and earth station node(s), link parameters and specifications, along with the service requirements and limitations. This paper presents efficient approach for accumulating the effective considerations that the designer should employ as a framework for building proper and efficient scheduling algorithm. The proposed approach provides proper solutions to the satellite communications impairments and satisfies the quality of service requirements in satellite communication networks.  


Author(s):  
Abdullah El-Haj ◽  
Shadi Aljawarneh

The existing research related to security mechanisms only focuses on securing the flow of information in the communication networks. There is a lack of work on improving the performance of networks to meet quality of service (QoS) constrains for various services. The security mechanisms work by encryption and decryption of the information, but do not consider the optimised use of the network resources. In this paper the authors propose a Secure Data Transmission Mechanism (SDTM) with Preemption Algorithm that combines between security and quality of service. Their developed SDTM enhanced with Malicious Packets Detection System (MPDS) which is a set of technologies and solutions. It enforces security policy and bandwidth compliance on all devices seeking to access Cloud network computing resources, in order to limit damage from emerging security threats and to allow network access only to compliant and trusted endpoint devices.


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