Packet loss

About: Packet loss is a research topic. Over the lifetime, 21235 publications have been published within this topic receiving 302453 citations.

IPTV Systems, Standards and Architectures: Part II - Application Layer FEC In IPTV Services

Abstract: Insufficient quality due to packet loss is one of the main obstacles to the broad adoption of IPTV services. However, with the use of end-to- end application layer reliability solutions IPTV services can be enabled over existing IP networks. In particular, application layer forward error correction (AL-FEC) solutions provide a straightforward and powerful means to overcome packet loss. Different solutions for different IPTV service types are introduced and compared. Selected performance results are provided to show the performance and potential of AL-FEC solutions for IPTV applications. From the results it is verified that even in non-perfect transmission conditions IPTV services can easily be deployed without requiring huge investments in new infrastructure and network equipment. Powerful AL-FEC enables high-quality IPTV services without abandoning the promises of the Internet: best effort transport and end-to-end reliability.

An Improved Stabilization Method for Sampled-Data Control Systems With Control Packet Loss

Abstract: This technical note presents a new method for stability analysis and stabilization of sampled-data systems with control packet loss. It is assumed that if the control packet from the controller to the actuator is lost, then the actuator input to the plant is set to zero. The new method is based on a novel construction of piecewise differentiable Lyapunov functionals by using an impulsive system representation of sampled-data systems. A significant feature of the new Lyapunov functionals is that they are continuous at impulse times but not necessarily positive definite inside the impulse intervals. Applying the new Lyapunov functionals to sampled-data systems with control packet loss, improved criteria for stability and stabilization are derived. The new criteria are proved theoretically to be less conservative than the existing results. Illustrative examples are given which substantiate the usefulness of the proposed method.

LRED: A Robust and Responsive AQM Algorithm Using Packet Loss Ratio Measurement

Abstract: Active queue management (AQM) is an effective means to enhance congestion control, and to achieve trade-off between link utilization and delay. The de facto standard, random early detection (RED), and many of its variants employ queue length as a congestion indicator to trigger packet dropping. Despite their simplicity, these approaches often suffer from unstable behaviors in a dynamic network. Adaptive parameter settings, though might solve the problem, remain difficult in such a complex system. Recent proposals based on analytical TCP control and AQM models suggest the use of both queue length and traffic input rate as congestion indicators, which effectively enhances stability. Their response time generally increases however, leading to frequent buffer overflow and emptiness. In this paper, we propose a novel AQM algorithm that achieves fast response time and yet good robustness. The algorithm, called Loss Ratio-based RED (LRED), measures the latest packet loss ratio, and uses it as a complement to queue length for adaptively adjusting the packet drop probability. We develop an analytical model for LRED, which demonstrates that LRED is responsive even if the number of TCP flows and their persisting times vary significantly. It also provides a general guideline for the parameter settings in LRED. The performance of LRED is further examined under various simulated network environments, and compared to existing AQM algorithms. Our simulation results show that, with comparable complexities, LRED achieves shorter response time and higher robustness. More importantly, it trades off the goodput with queue length better than existing algorithms, enabling flexible system configurations

Method of preserving data packet sequencing

Abstract: A method of preserving data packet sequencing between nodes in a radio telecommunications network in which the User Datagram Protocol/Internet Protocol (UDP/IP) is utilized to communicate between the nodes of the network Sequencing numbers are added to the headers of a plurality of data packets which are transmitted over a plurality of data transmission paths The UDP/IP protocol is adapted with a resequencing layer which is utilized at the receiving node to resequence the packets

Multimedia streaming using multiple TCP connections

Abstract: In recent years, multimedia applications over the Internet become increasingly popular. However, packet loss, delay, and time-varying bandwidth of the Internet have remained the major problems for multimedia streaming applications. As such, a number of approaches, including network infrastructure and protocol, source and channel coding, have been proposed to either overcome or alleviate these drawbacks of the Internet. In this article, we propose the MultiTCP system, a receiver-driven, TCP-based system for multimedia streaming over the Internet. Our proposed algorithm aims at providing resilience against short term insufficient bandwidth by using multiple TCP connections for the same application. Our proposed system enables the application to achieve and control the desired sending rate during congested periods, which cannot be achieved using traditional TCP. Finally, our proposed system is implemented at the application layer, and hence, no kernel modification to TCP is necessary. We analyze the proposed system, and present simulation and experimental results to demonstrate its advantages over the traditional single-TCP-based approach.