Transport Layer Mechanisms Research Articles

Remote Production for Live Holographic Teleportation Applications in 5G Networks

Holographic Teleportation is an emerging media application allowing people or objects to be teleported in a real-time and immersive fashion into the virtual space of the audience side. Compared to the traditional video content, the network requirements for supporting such applications will be much more challenging. In this paper, we present a 5G edge computing framework for enabling remote production functions for live holographic Teleportation applications. The key idea is to offload complex holographic content production functions from end user premises to the 5G mobile edge in order to substantially reduce the cost of running such applications on the user side. We comprehensively evaluated how specific network-oriented and application-oriented factors may affect the performances of remote production operations based on 5G systems. Specifically, we tested the application performance from the following four dimensions: (1) different data rate requirements with multiple content resolution levels, (2) different transport-layer mechanisms over 5G uplink radio, (3) different indoor/outdoor location environments with imperfect 5G connections and (4) different object capturing scenarios including the number of teleported objects and the number of sensor cameras required. Based on these evaluations we derive useful guidelines and policies for future remote production operation for holographic Teleportation through 5G systems.

IDDTL: A Novel Identified Internet Data Transport Layer Mechanism

This paper proposes an identified data transport layer (IDDTL) mechanism, which is implemented based on our present concepts of connection identifier (CID) and CID additional information (CID-info). Since the fast evolved Internet scale and largely emerging various applications, especially with the new Internet architectures developed such as information centric network (ICN), the traditional end-to-end transport model has been exposed many defects in sorts of aspects, such as network management, flexibility and security. The novelty of the mechanism consists of two points: 1) it always conceals part of the communication information during the specific data transport process; 2) it splits the whole end-to-end communication process into two segments and forms a three-party and two-segment communication process model. Performance analysis shows that the mechanism could easily mitigate the problems such as distributed denial of service (D/DOS) attacks and greatly improve the network management, flexibility and mobility. Furthermore, our simulation and test results demonstrate that IDDTL can be implemented with unique identifiers within an acceptable extra time cost of about 3.6 useconds compared with the traditional end-to-end model.

A new asymmetric spectrum assignment method to improve spectrum efficiency for spectrum-sliced optical network

Bandwidth flexible optical network, referred to as spectrum-sliced elastic optical path network (SLICE), is a promising solution to achieve both high data rate transport and high spectrum-efficient. However, current methods to solve Routing and Spectrum Assignment (RSA) problems for SLICE network, conservatively accept the constraint that spectrum resources should be assigned symmetrically in both directions of one optical path. Since most data traffics are asymmetric by nature, the conventional way to simply aggregate them to form symmetric flows for transport layer is not as flexible as offering a direct asymmetric transport layer mechanism. In this paper, we present a novel Asymmetric Spectrum Assignment (ASA) method to future improve spectrum efficiency by saving redundant spectrum resources for the lighter load direction of one optical path to accommodate more incoming connection requests. We describe the concept of ASA and the enabling technologies to implement it. To compare the performance of proposed ASA with conventional Symmetric Spectrum Assignment (SSA), we classify the evaluations into four scenarios. We also observe the effect of various grid width values. Simulation result shows that: (i) ASA method always significantly outperforms the SSA method in any situation. (ii) Routing asymmetry is not necessary needed and spectrum assignment optimization takes obvious effect only under the situation of “random slot” grid width value.

Beyond network simulators: Fostering novel distributed applications and protocols through extendible design

Simulation has been of paramount importance to the development of novel Internet protocols. Such an approach typically focuses on one of three domains: wireless and other link-layer technologies, routing protocols, and transport-layer mechanisms and protocols. Existing techniques can tackle well simulation at layers 2, 3 and 4 of the TCP/IP architecture, but are not flexible enough to appropriately deal with application-layer protocols. These require simulators that support the modeling of networks and components with different levels of abstraction. Simmcast is an object-oriented framework that focuses on the necessary flexibility for application-layer protocol research. A simulation can be developed by the simple extension of building blocks that closely resemble components of a real network such as hosts, links and routers. The internal complexity of these components, however, is hidden from the user, so he/she can focus on the implementation of the desired protocol characteristics. This paper describes the flexible simulation architecture proposed and instantiated through Simmcast, and draws lessons from our experience in designing, implementing and deploying it. We also present framework instances used to evaluate application-layer protocols, exemplifying how different kinds of simulations can be developed with Simmcast.

A delay‐based transport layer mechanism for fair TCP throughput over 802.11 multihop wireless mesh networks

AbstractCities worldwide have planned and implemented large‐scale wireless mesh network (WMN) deployments. These mesh deployments are expected to provide broadband mobile access to the Internet at a low cost to the user. This paper considers the fairness problem affecting nodes in multihop WMNs. Many existing approaches to coping with the fairness problem are unsuitable because they necessitate modifications to the IEEE 802.11 MAC protocol or node queueing operations. Such modifications require a change in hardware/firmware at every node in the WMN. Thus, these approaches are less favorable from the points of view of interoperability and cost. Without modifying the lower layer protocols, this study identifies TCP parameters that impact throughput fairness and proposes an adjustment to these parameters to reduce frame collisions and to improve throughput fairness. Using simple mathematical formulations and ns2 simulations, this study shows that the frame transmissions from each node can be effectively controlled by properly controlling the delayed ACK timer and by using a suitable advertised window. The proposed method, in addition to fairness, requires fewer buffer resources than other methods. Moreover, it is not sensitive to the carrier sense range. It is also simple and easy to deploy. Copyright © 2011 John Wiley & Sons, Ltd.

Adaptive end-to-end QoS for multimedia over heterogeneous wireless networks

There has been a surge of interest for multimedia applications over wireless networks in recent years. A colossal number of ways have been proposed to decrease delay, delay jitter and loss in wireless networks and good user-perceived quality in video over internet. This paper studies the multimedia over heterogeneous wireless networks, requirements and problems, and proposes a new scheme to overcome the obstacles. The proposed scheme, takes into account the effects of Application-Level Wireless Multilevel ECN marking (AWMECN), thus helps us overcome the congestion/loss mistake problems. For handoff, handover and lossy link problems, it is considered that a freezing mechanism is in use in application layer and assumed that the upper layers can be aware of disconnection periods to make the rate adaptation decisions. Also a new scheme has been added to receiver to gracefully degrade the quality when no other action is available to combat the long delays without data which is caused by handoffs and wireless temporary disconnections. The transport layer mechanism is chosen to be UDP for avoiding TCP regarding problems. We believe that obtaining a good quality of video depends on good performance of all layers and tried to use the best mechanism in each layer.

Study and performance analysis of transport layer mechanisms applied in military radio environment

The need for reliable data communication performed in critical conditions and the respect of real-time constraints is an open issue in radio-military networks, where fixed and nomadic terminals communicate by means of tactical radio infrastructures. In this perspective the need for real-time constraints plays a fundamental role when the interested applications are remote control, radar traces acquisition and command/control message exchanging. In this paper, the difficulties of data transportation in wireless networks are considered by taking emphasis on the TCP-NewReno implementation that does not guarantee satisfying performance results in such environments. The whole investigation is addressed at designing and validating novel transmission mechanisms to be implemented at the transport layer, directly inherited from standard TCP transmission scheme. TCP-radio and TCP-radio-newRTO proposals are thus discussed and compared with TCP-Westwood+ and TCP-NewReno in order to highlight the necessity for an enhanced transport protocol for data communication performed in radio military environments, where the data transfer is heavily affected by hazardous conditions, such as scattering, shadowing and slow fading. All the tests have been performed through the employment of a network emulator tuned to the radio characteristics in terms of end-to-end delay, available bandwidth and channel modelling.

Improving TCP Performance in Mobile Networks

In this letter, a new transport layer mechanism is proposed to improve the performance of transport control protocol (TCP) in mobile networks. The proposed mechanism is comprised of two parts: a loss classifier (LC) and a congestion window extrapolator (CWE). Based on LC, the cause of packet loss during roaming is determined. If the loss is considered to be caused by congestion in the wireline, the congestion window is halved; otherwise, the packet is considered to be lost in the last hop, the wireless portion, and the sender adjusts the size of the congestion window based on CWE. We conduct simulations to evaluate the performance of the proposed mechanism. The results show that our mechanism significantly improves TCP performance as compared with existing solutions for mobile networks.

Analysis of TCP Performance Under Joint Rate and Power Adaptation in Cellular WCDMA Networks

To improve the spectral efficiency while meeting the radio link level quality of service requirements such as the bit-error-rate (BER) requirements for the different wireless services, transmission rate and power corresponding to the different mobile users can be dynamically varied in a cellular wideband code-division multiple-access (WCDMA) network depending on the variations in channel interference and fading conditions. This paper models and analyzes the performance of transmission control protocol (TCP) under joint rate and power adaptation with constrained BER requirements for downlink data transmission in a cellular variable spreading factor (VSF) WCDMA network. The aim of this multilayer modeling of the WCDMA radio interface is to better understand the interlayer protocol interactions and identify suitable transport and radio link layer mechanisms to improve TCP performance in a wide-area cellular WCDMA network.

Providing Differentiated Service to TCP Flows Over Bandwidth on Demand Geostationary Satellite Networks

The elasticity of transmission control protocol (TCP) traffic complicates attempts to provide performance guarantees to TCP flows. The existence of different types of networks and environments on the connections' paths only aggravates this problem. In this paper, simulation is the primary means for investigating the specific problem in the context of bandwidth on demand (BoD) geostationary satellite networks. Proposed transport-layer options and mechanisms for TCP performance enhancement, studied in the single connection case or without taking into account the media access control (MAC)-shared nature of the satellite link, are evaluated within a BoD-aware satellite simulation environment. Available capabilities at MAC layer, enabling the provision of differentiated service to TCP flows, are demonstrated and the conditions under which they perform efficiently are investigated. The BoD scheduling algorithm and the policy regarding spare capacity distribution are two MAC-layer mechanisms that appear to be complementary in this context; the former is effective at high levels of traffic load, whereas the latter drives the differentiation at low traffic load. When coupled with transport layer mechanisms they can form distinct bearer services over the satellite network that increase the differentiation robustness against the TCP bias against connections with long round-trip times. We also explore the use of analytical, fixed-point methods to predict the performance at transport level and link level. The applicability of the approach is mainly limited by the lack of analytical models accounting for prioritization mechanisms at the MAC layer and the nonuniform distribution of traffic load among satellite terminals.

Application of H.263+ Video Coding Modes in Lossy Packet Network Environments

The quality of real-time audio and video information transmitted via today's Internet suffers severely from often significant packet losses. While this problem is well understood and solved for existing audio coding schemes, support from the video coding standards themselves is required for video streams. This paper presents the newly introduced error resilience mechanisms built into the second version of H.263 (1998), known under its working name H.263+, and addresses the corresponding packetization format issues that together significantly improve the image quality at packet loss rates up to 20%. In particular, it is support from the video coding algorithm itself, paired with appropriate transport layer mechanisms, that leads to significant improvements of perceived image quality for communicative as well as retrieval applications at moderate bit rates up to some 100 kbit/s.