Adaptive Early Packet Discarding Scheme to Improve Network Delay Characteristics of Real-Time Flows (original) (raw)
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Improving Delay Characteristics of Real-Time Flows by Adaptive Early Packet Discarding
Lecture Notes in Computer Science, 2006
The quality of real-time applications is significantly affected by the delay of packets traversing a network. Some real-time applications set limits for acceptable network delay, and thus, a packet delayed longer than this limit before arriving at its destination is not only worthless but also harmful to the quality of the application because it may increase the queuing delay of other packets. Therefore, we propose an adaptive scheme for real-time applications in which such packets are discarded early. In this scheme, packets experiencing too much delay are discarded at intermediate nodes based on the delay limit for the application and the delay experienced by each packet. Such early discarding of packets is expected to improve the overall delay characteristics of real-time flows competing for network resources shared only by those flows. Simulation results showed that our scheme is effective.
The delay-friendliness of TCP for real-time traffic
IEEE/ACM Transactions on Networking, 2010
TCP has traditionally been considered inappropriate for real-time applications. Nonetheless, popular applications such as Skype use TCP since UDP packets cannot pass through restrictive network address translators (NATs) and firewalls. Motivated by this observation, we study the delay performance of TCP for real-time media flows. We develop an analytical performance model for the delay of TCP. We use extensive experiments to validate the model and to evaluate the impact of various TCP mechanisms on its delay performance. Based on our results, we derive the working region for VoIP and live video streaming applications and provide guidelines for delay-friendly TCP settings. Our research indicates that simple application-level schemes, such as packet splitting and parallel connections, can reduce the delay of real-time TCP flows by as much as 30% and 90%, respectively.
IEEE/ACM Transactions on Networking, 2005
Currently there is no control for real-time traffic sources in IP networks. This is a serious problem because real-time traffic can not only congest the network but can also cause unfairness and starvation of TCP traffic. However, it is not possible to apply current solutions for Internet to the networks with high bandwidth-delay products and high bit error rates. The channel errors may result in inaccurate congestion control decisions and unnecessary rate throttles leading to severe performance degradation. This problem is amplified in the links with high bandwidth-delay products, since the link is inefficiently utilized for a very long time until the unnecessary rate throttle is recovered. In this paper, a new Rate Control Scheme, RCS, is introduced for real-time interactive applications in networks with high bandwidth-delay products and high bit error rates. RCS is based on the concept of using dummy packets to probe the availability of network resources. Dummy packets are treated as low priority packets and consequently they do not affect the throughput of actual data traffic. Therefore, RCS requires all the routers in the connection path to support some priority policy. A new algorithm is also proposed to improve the robustness of the RCS to temporal signal loss conditions. The delay-bound considerations for real-time traffic sources using RCS rate control scheme are also investigated. Simulation experiments show that in environments with high bandwidth-delay products and high bit error rates, RCS achieves high throughput performance without penalizing TCP connections. Index Terms-Flow control, high bandwidth-delay products, high bit error rates, real-time protocols.
V-NET: A versatile network architecture for flexible delay guarantees in real-time networks
IEEE Transactions on Computers, 2000
AbstractÐThis paper proposes a Versatile Network Architecture (-NET) to support flexible delay guarantees for applications in realtime networks. Applications communicate over the -NET by using end-to-end network connections which support real-time and reliability characteristics tailored to meet the application's specified requirements. -NET differs from other proposed architectures in that, in addition to addressing the issue of quality of service (QoS) feasibility for a wide spectrum of real-time applications, it also provides a mechanism to determine a network state dependent range of feasible delay values at each switching node along the routing path. These delay ranges can be used to assign per-node delays that reflect the resource availability of the node, thereby reducing the likelihood of bottlenecks along the routing path. The -NET delay guarantees are provided for a variety of packet scheduling algorithms and traffic policing mechanisms. This flexibility is an important design consideration as a real-time network architecture must accommodate existing and future multimedia applications, with hard-, soft-, and non-real-time traffic. The performance evaluation results demonstrate the efficiency of this scheme in handling different traffic scenarios and QoS requirements. We have shown that it is possible, and indeed efficient, to determine an upper-bound on the delay of real-time traffic, when using our per-node delay assignment policies. Fig. 2. Switching node architecture.
Controlling queuing delays for real-time communication
ACM SIGCOMM Computer Communication Review, 2018
Real-time media communication requires not only congestion control, but also minimization of queuing delays to provide interactivity. In this work we consider the case of real-time communication between web browsers (WebRTC) and we focus on the interplay of an end-to-end delay-based congestion control algorithm, i.e. the Google congestion control (GCC), with two delay-based AQM algorithms, namely CoDel and PIE, and two flow queuing schedulers, i.e. SFQ and Fq_Codel. Experimental investigations show that, when only GCC flows are considered, the end-to-end algorithm is able to contain queuing delays without AQMs. Moreover the interplay of GCC flows with PIE or CoDel leads to higher packet losses with respect to the case of a DropTail queue. In the presence of concurrent TCP traffic, PIE and CoDel reduce the queuing delays with respect to DropTail at the cost of increased packet losses. In this scenario flow queuing schedulers offer a better solution.
Supporting excess real-time traffic with active drop queue
IEEE/ACM Transactions on Networking, 2006
Real-time applications often stand to benefit from service guarantees, and in particular delay guarantees. However, most mechanisms that provide delay guarantees also hard-limit the amount of traffic the application can generate, i.e., to enforce to a traffic contract. This can be a significant constraint and interfere with the operation of many real-time applications. Our purpose in this paper is to propose and investigate solutions that overcome this limitation.We have four major goals: (1) guarantee a delay bound to a contracted amount of real-time traffic; (2) transmit with the same delay bound as many excess real-time packets as possible; (3) enforce a given link sharing ratio between excess real-time traffic and other service classes, e.g., best-effort; (4) preserve the ordering of real-time packets, if required. Our approach is based on a combination of buffer management and scheduling mechanisms for both guaranteeing delay bounds, while allowing the transmission of excess traffic. We evaluate the "cost" of our scheme by measuring the processing overhead of an actual implementation, and we investigate its performance by means of simulations using video traffic traces.
A Study on Delay Based Congestion Control Challenges
2015
Bufferbloat" is the reason for delay bloat problem and also main reason for congestion. It is basically the presence of large, unmanaged network buffers, primarily across the edge devices of the Internet. Overly large buffers have been placed in some models of equipment by their manufacturers. In this equipment bufferbloat occurs when a network link becomes congested, causing packets to become queued for too long in those buffers. Tcptrace and xplot.org is useful tool for looking at the bufferbloat problem.Delay based approach is very popular among congestion control approaches because of its low latency characteristics as study reveal , but our paper wants to claims a valuable direction. Although delay-based congestion control protocols promise to deliver better performance than loss based congestion control protocols, they have not yet been widely incorporated to the Internet. It is not easy yet; there are many challenges which should be overcome for the quest of low delay.
ACM SIGMETRICS Performance Evaluation Review, 2008
TCP has traditionally been considered unfriendly for realtime applications. Nonetheless, popular applications such as Skype use TCP since UDP packets cannot pass through many NATs and firewalls. Motivated by this observation, we study the delay performance of TCP for real-time media flows. We develop an analytical performance model for the delay of TCP. We use extensive experiments to validate the model and to evaluate the impact of various TCP mechanisms on its delay performance. Based on our results, we derive the working region for VoIP and live video streaming applications and provide guidelines for delay-friendly TCP settings. Our research indicates that simple application-level schemes, such as packet splitting and parallel connections, can reduce the delay of real-time TCP flows by as much as 30% and 90%, respectively.