Adaptation strategies for MGS scalable video streaming

An adaptive streaming framework consists of a video codec that can produce video encoded at a variety of rates, a transport protocol that supports an effective rate/congestion control mechanism, and an adaptation strategy in order to match the video source rate to the available network throughput. The main parameters of the adaptation strategy are encoder configuration, video extraction method, determination of video extraction rate, send rate control, retransmission of lost packets, decoder buffer status, and packetization method. This paper proposes optimal adaptation strategies, in terms of received video quality and used network resources, at the codec and network levels using a medium grain scalable (MGS) video codec and two transport protocols with built-in congestion control, TCP and DCCP. Key recommendations are presented to obtain the best results in adaptive video streaming using TCP or DCCP based on extensive experimental results over the Internet.     

Randomly Ranked Mini Slots for Fair and Efficient Medium Access Control in Ad Hoc Networks

Ad hoc networks offer infrastructure-free operation, where no entity can provide reliable coordination among nodes. Medium access Control (MAC) protocols in such a network must overcome the inherent unreliability of the network and provide high throughput and adequate fairness to the different flows of traffic. In this paper, we propose a MAC protocol that can achieve an excellent balance between throughput and fairness. Our protocol has two versions: randomly ranked mini slots (RRMS) utilizes control-message handshakes similar to IEEE 802.11. Randomly ranked mini slots with busy tone (RRMS-BT) is the better performer of the two, but requires a receiver busy tone. The protocol makes use of granule time slots and sequences of pseudorandom numbers to maximize spatial reuse and divide the throughput fairly among nodes. We demonstrate the performance of this protocol using simulation with fixed and random topologies and show that these results are robust to difficult network configurations and unsynchronized clocks. We further develop novel metrics of long-term and short-term fairness for rigorous performance evaluation. Our simulation results include a detailed comparison between the proposed protocol and existing protocols that have been shown to excel in terms of throughput or fairness     

End-to-end differentiation of congestion and wireless losses

In this paper, we explore end-to-end loss differentiation algorithms (LDAs) for use with congestion-sensitive video transport protocols for networks with either backbone or last-hop wireless links. As our basic video transport protocol, we use UDP in conjunction with a congestion control mechanism extended with an LDA. For congestion control, we use the TCP-Friendly Rate Control (TFRC) algorithm. We extend TFRC to use an LDA when a connection uses at least one wireless link in the path between the sender and receiver. We then evaluate various LDAs under different wireless network topologies, competing traffic, and fairness scenarios to determine their effectiveness. In addition to evaluating LDAs derived from previous work, we also propose and evaluate a new LDA, ZigZag, and a hybrid LDA, ZBS, that selects among base LDAs depending upon observed network conditions. We evaluate these LDAs via simulation, and find that no single base algorithm performs well across all topologies and competition. However, the hybrid algorithm performs well across topologies and competition, and in some cases exceeds the performance of the best base LDA for a given scenario. All of the LDAs are reasonably fair when competing with TCP, and their fairness among flows using the same LDA depends on the network topology. In general, ZigZag and the hybrid algorithm are the fairest among all LDAs.     

TCP-friendly flow control of wireless multimedia using ECN marking

In a wireless network packet losses can be caused not only by network congestion but also by unreliable error-prone wireless links. Therefore, flow control schemes which use packet loss as a congestion measure cannot be directly applicable to a wireless network because there is no way to distinguish congestion losses from wireless losses. In this paper, we extend the so-called TCP-friendly flow control scheme, which was originally developed for the flow control of multimedia flows in a wired IP network environment, to a wireless environment. The main idea behind our scheme is that by using explicit congestion notification (ECN) marking in conjunction with random early detection (RED) queue management scheme intelligently, it is possible that not only the degree of network congestion is notified to multimedia sources explicitly in the form of ECN-marked packet probability but also wireless losses are hidden from multimedia sources. We calculate TCP-friendly rate based on ECN-marked packet probability instead of packet loss probability, thereby effectively eliminating the effect of wireless losses in flow control and thus preventing throughput degradation of multimedia flows travelling through wireless links. In addition, we refine the well-known TCP throughput model which establishes TCP-friendliness of multimedia flows in a way that the refined model provides more accurate throughput estimate of a TCP flow particularly when the number of TCP flows sharing a bottleneck link increases. Through extensive simulations, we show that the proposed scheme indeed improves the quality of the delivered video significantly while maintaining TCP-friendliness in a wireless environment for the case of wireless MPEG-4 video.     

TFRC friendliness and the case of ECN

The TFRC protocol has been proposed as a TCP‐friendly protocol to transport streaming media over the Internet. However, its deployment is still questionable because it has not been compared to other important protocols, analysed in the presence of important mechanisms, such as the explicit congestion notification (ECN), and studied under more realistic network conditions. In this paper, we address these three aspects, including other congestion control protocols not considered before in the same investigation, such as TCP Tahoe, Reno, Newreno, Vegas, Sack, GAIMD, and the Binomial algorithms, the effect of using ECN in the friendliness of the protocols, and the fairness of the protocols under static and dynamic network conditions. We found that TFRC can be safely deployed in the Internet if competing with TCP Tahoe, New Reno and SACK since fairness is achieved under all scenarios considered. We also found that ECN actually helps in achieving better fairness. However, fairness problems arise when TFRC competes with TCP Reno, GAIMD, SQRT or IIAD in static or dynamic conditions, or both. We used normalized throughput, fairness index, and convergence time as the main performance metrics for comparison. Copyright © 2004 John Wiley & Sons, Ltd.     

Dual-Resource TCP/AQM for Processing-Constrained Networks

This paper examines congestion control issues for TCP flows that require in-network processing on the fly in network elements such as gateways, proxies, firewalls and even routers. Applications of these flows are increasingly abundant in the future as the Internet evolves. Since these flows require use of CPUs in network elements, both bandwidth and CPU resources can be a bottleneck and thus congestion control must deal with ldquocongestionrdquo on both of these resources. In this paper, we show that conventional TCP/AQM schemes can significantly lose throughput and suffer harmful unfairness in this environment, particularly when CPU cycles become more scarce (which is likely the trend given the recent explosive growth rate of bandwidth). As a solution to this problem, we establish a notion of dual-resource proportional fairness and propose an AQM scheme, called Dual-Resource Queue (DRQ), that can closely approximate proportional fairness for TCP Reno sources with in-network processing requirements. DRQ is scalable because it does not maintain per-flow states while minimizing communication among different resource queues, and is also incrementally deployable because of no required change in TCP stacks. The simulation study shows that DRQ approximates proportional fairness without much implementation cost and even an incremental deployment of DRQ at the edge of the Internet improves the fairness and throughput of these TCP flows. Our work is at its early stage and might lead to an interesting development in congestion control research.     

区分服务中一种拥塞感知的单速三色标记算法

确保服务的实现依赖于在边界路由器执行的数据包标记策略和在核心路由器执行的队列管理策略.TCP流由于其拥塞自适应的特点对丢包很敏感,网络拥塞对其吞吐量影响很大.为此,我们设计了一种拥塞感知的单速三色标记算法CASR3CM.仿真实验表明,该算法不仅提高了AS TCP流的平均吞吐量,而且增强了吞吐量的稳定性.另外该算法也提高了AS TCP流之间占用带宽的公平性.     

Video-aware opportunistic network coding over wireless networks

In this paper, we study video streaming over wireless networks with network coding capabilities. We build upon recent work, which demonstrated that network coding can increase throughput over a broadcast medium, by mixing packets from different flows into a single packet, thus increasing the information content per transmission. Our key insight is that, when the transmitted flows are video streams, network codes should be selected so as to maximize not only the network throughput but also the video quality. We propose video-aware opportunistic network coding schemes that take into account both the decodability of network codes by several receivers and the importance and deadlines of video packets. Simulation results show that our schemes significantly improve both video quality and throughput. This work is a first step towards content-aware network coding.     

An early congestion feedback and rate adjustment schemes for many-to-one communication in cloud-based data center networks

Cloud data centers are playing an important role for providing many online services such as web search, cloud computing and back-end computations such as MapReduce and BigTable. In data center network, there are three basic requirements for the data center transport protocol such as high throughput, low latency and high burst tolerance. Unfortunately, conventional TCP protocols are unable to meet the requirements of data center transport protocol. One of the main practical issues of great importance is TCP Incast to occur many-to-one communication sessions in data centers, in which TCP experiences sharp degradation of throughput and higher delay. This important issue in data center networks has already attracted the researchers because of the development of cloud computing. Recently, few solutions have been proposed for improving the performance of TCP in data center networks. Among that, DCTCP is the most popular protocol in academic as well as industry areas due to its better performance in terms of throughput and latency. Although DCTCP provides significant performance improvements, there are still some defects in maintaining the queue length and throughput when the number of servers is too large. To address this problem, we propose a simple and efficient TCP protocol, namely NewDCTCP as an improvement of DCTCP in data center networks. NewDCTCP modified the congestion feedback and window adjusting schemes of DCTCP to mitigate the TCP Incast problem. Through detailed QualNet experiments, we show that NewDCTCP significantly outperforms DCTCP and TCP in terms of goodput and latency. The experimental results also demonstrate that NewDCTCP flows provide better link efficiency and fairness with respect to DCTCP.     

A dynamic load-aware congestion control scheme in optical burst switching networks

The most important design goal in Optical Burst Switching (OBS) networks is to reduce burst loss resulting from resource contention. Especially, the higher the congestion degree in the network is, the higher the burst loss rate becomes. The burst loss performance can be improved by employing an appropriate congestion control. In this paper, to actively avoid contentions, we propose a dynamic load-aware congestion control scheme that operates based on the highest (called ‘peak load’) of the loads of all links over the path between each pair of ingress and egress nodes in an OBS network. We also propose an algorithm that dynamically determines a load threshold for adjusting burst sending rate, according to the traffic load in a network. Further, a simple signalling method is developed for our proposed congestion control scheme. The proposed scheme aims to (1) reduce the burst loss rate in OBS networks and (2) maintain reasonable throughput and fairness. Simulation results show that the proposed scheme reduces the burst loss rate significantly, compared to existing OBS protocols (with and without congestion control), while maintaining reasonable throughput and fairness. Simulation results also show that our scheme keeps signalling overhead due to congestion control at a low level.     

Improving Transport Layer Performance in Multihop Ad Hoc Networks by Exploiting MAC Layer Information

The traditional TCP congestion control mechanism encounters a number of new problems and suffers a poor performance when the IEEE 802.11 MAC protocol is used in multihop ad hoc networks. Many of the problems result from medium contention at the MAC layer. In this paper, we first illustrate that severe medium contention and congestion are intimately coupled, and TCP's congestion control algorithm becomes too coarse in its granularity, causing throughput instability and excessively long delay. Further, we illustrate TCP's severe unfairness problem due to the medium contention and the tradeoff between aggregate throughput and fairness. Then, based on the novel use of channel busyness ratio, a more accurate metric to characterize the network utilization and congestion status, we propose a new wireless congestion control protocol (WCCP) to efficiently and fairly support the transport service in multihop ad hoc networks. In this protocol, each forwarding node along a traffic flow exercises the inter-node and intra-node fair resource allocation and determines the MAC layer feedback accordingly. The end-to-end feedback, which is ultimately determined by the bottleneck node along the flow, is carried back to the source to control its sending rate. Extensive simulations show that WCCP significantly outperforms traditional TCP in terms of channel utilization, delay, and fairness, and eliminates the starvation problem     

A Link Adaptive Transport Protocol for Multimedia Streaming Applications in Multi Hop Wireless Networks

Transport layer performance in multi hop wireless networks has been greatly challenged by the intrinsic characteristics of these networks. In particular, the nature of congestion, which is mainly due to medium contention in multi hop wireless networks, challenges the performance of traditional transport protocols in such networks. In this paper, we first study the impact of medium contention on transport layer performance and then propose a new transport protocol for improving quality of service performance in multi hop wireless networks. Our proposed protocol, Link Adaptive Transport Protocol provides a systemic way of controlling transport layer offered load for multimedia streaming applications, based on the degree of medium contention information received from the network. Simulation results show that the proposed protocol provides an efficient scheme to improve quality of service performance metrics, such as end-to-end delay, jitter, packet loss rate, throughput smoothness and fairness for media streaming applications. In addition, our scheme requires few overhead and does not maintain any per-flow state table at intermediate nodes. This makes it less complex and more cost effective.     

A framework for delivering fine granularity and fair service discrimination in DiffServ networks

In order to achieve a quality of service (QoS) capable of satisfying an ever increasing range of user requirements, differentiated services (DiffServ) have been introduced as a scalable solution that emerges ‘naturally’ from today's best effort service approach. Mapping the packet treatment into a small number of per hop behaviours (PHBs) is the key idea behind the scalability of DiffServ but this comes at the cost of loosing some behavioural differentiation and some fairness between flows multiplexed into the same aggregated traffic. The paper proposes a novel simple and effective DiffServ approach, the ‘Simple Weighted Integration of diFferentiated Traffic’ (SWIFT), and uses it in a series of simulations covering a relatively wide range of local network conditions. Measured voice and video traffic traces and computer generated self‐similar background traffic were used in simulations performed at various congestion levels and for in‐profile and out‐of‐profile source behaviour. The resulted throughput, mean delay, maximum delay and jitter are used to asses SWIFT's capabilities—isolation of the in‐profile traffic from congestion effects, treatment differentiation, increased resource utilization, fairness in treatment under congestion, and incentivity for nice behaviour. Comparisons with other approaches employing traffic control are also provided. Copyright © 2003 John Wiley & Sons, Ltd.     

The Impact of Loss Recovery on Congestion Control for Reliable Multicast

Most existing reliable multicast congestion control (RMCC) mechanisms try to emulate TCP congestion control behaviors for achieving TCP-compatibility. However, different loss recovery mechanisms employed in reliable multicast protocols, especially NAK-based retransmission and local loss recovery mechanisms, may lead to different behaviors and performance of congestion control. As a result, reliable multicast flows might be identified and treated as non-TCP-friendly by routers in the network. It is essential to understand those influences and take them into account in the development and deployment of reliable multicast services. In this paper, we study the influences comprehensively through analysis, modelling and simulations. We demonstrate that NAK-based retransmission and/or local loss recovery mechanisms are much more robust and efficient in recovering from single or multiple packet losses within a single round-trip time (RTT). For a better understanding on the impact of loss recovery on RMCC, we derive expressions for steady-state throughput of NAK-based RMCC schemes, which clearly brings out the throughput advantages of NAK-based RMCC over TCP Reno. We also show that timeout effects have little impact on shaping the performance of NAK-based RMCC schemes except for extremely high loss rates (>0.2). Finally, we use simulations to validate our findings and show that local loss recovery may further increase the throughput and deteriorate the fairness properties of NAK-based RMCC schemes. These findings and insights could provide useful recommendations for the design, testing and deployment of reliable multicast protocols and services     

PASA: power adaptation for starvation avoidance to deliver wireless multimedia

In recent years, there has been an increasing interest to deliver multimedia services over wireless ad hoc networks. Due to the existence of hidden terminal and absence of central control, the medium access control protocol as used in the ad hoc networks may lead to channel capture, where some flows monopolize the channel while others suffer from starvation. As a consequence, the system throughput and fairness are greatly degraded. After showing that static power control leads to channel capture, we propose and study a distributed dynamic power control scheme termed "power adaptation for starvation avoidance" (PASA), which dynamically adjusts the transmission power of a node so as to avoid starvation. PASA is shown to be effective in breaking channel captures, hence improving short-term fairness among contending flows. It is simple, fully autonomous and requires no communication overhead. Via extensive simulations, we show that our power control algorithm achieves much better fairness without compromising system throughput through better spatial reuse. Our experiments with video sequences transmitting over different network topologies show that PASA achieves much better video quality with lower start-up delay and buffer requirement.     

FCA-ItswTCM:区分服务中一种公平的拥塞自适应标记算法

 本文提出了一种新的滑动窗口标记算法——公平的拥塞自适应标记算法(FCA-ItswTCM).算法近似识别TCP流和UDP流,适度区分标记,规避拥塞控制机制对公平性的影响;细粒度描述拥塞,预测拥塞,以此自适应调节各流注入黄包比例,兼顾网络拥塞状态对公平性的影响.仿真实验表明,与其他几种滑动窗口标记算法相比,FCA-ItswTCM对确保TCP流和UDP流带宽共享的公平性、提高资源利用率及系统稳定性有较好的效果.     

A study of a simple preventive transport protocol

The main qualities of a protocol for multimedia flows transportation are related to the way congestions are handled. This paper addresses the problem of end-to-end congestion control performed in the Internet transport layer. We present a simple protocol called Primo, which determines the appropriate sending rate in order to maximize network resources usage and minimize packets loss. Comparison with existing transport protocols (Tcp Reno, Sack, Vegas andTfrc) are considered, regarding various efficiency criteria such as sending and reception rates stability, loss rate, resources occupancy rate and fairness.     

Multicasting with Localized Control in Wireless Ad Hoc Networks

This paper investigates how to support multicasting in wireless ad hoc networks without throttling the dominant unicast flows. Unicast flows are usually congestion-controlled with protocols like TCP. However, there are no such protocols for multicast flows in wireless ad hoc networks and multicast flows can therefore cause severe congestion and throttle TCP-like flows in these environments. Based on a cross-layer approach, this paper proposes a completely-localized scheme to prevent multicast flows from causing severe congestion and the associated deleterious effects on other flows in wireless ad hoc networks. The proposed scheme combines the layered multicast concept with the routing-based congestion avoidance idea to reduce the aggregated rate of multicast flows when they use excessive bandwidth on a wireless link. Our analysis and extensive simulations show that the fully-localized scheme proposed in this paper is effective in ensuring the fairness of bandwidth sharing between multicast and unicast flows in wireless ad hoc networks.     

End-to-end optimized TCP-friendly rate control for real-time video streaming over wireless multi-hop networks

Rate control is an important issue in video streaming applications. The most popular rate control scheme over wired networks is TCP-Friendly Rate Control (TFRC), which is designed to provide optimal transport service for unicast multimedia delivery based on the TCP Reno’s throughput equation. It assumes perfect link quality, treating network congestion as the only reason for packet losses. Therefore, when used in wireless environment, it suffers significant performance degradation because of packet losses arising from time-varying link quality. Most current research focuses on enhancing the TFRC protocol itself, ignoring the tightly coupled relation between the transport layer and other network layers. In this paper, we propose a new approach to address this problem, integrating TFRC with the application layer and the physical layer to form a holistic design for real-time video streaming over wireless multi-hop networks. The proposed approach can achieve the best user-perceived video quality by jointly optimizing system parameters residing in different network layers, including real-time video coding parameters at the application layer, packet sending rate at the transport layer, and modulation and coding scheme at the physical layer. The problem is formulated and solved as to find the optimal combination of parameters to minimize the end-to-end expected video distortion constrained by a given video playback delay, or to minimize the video playback delay constrained by a given end-to-end video distortion. Experimental results have validated 2–4 dB PSNR performance gain of the proposed approach in wireless multi-hop networks by using H.264/AVC and NS-2.     

Counter-intuitive throughput behaviors in networks under end-to-end control

It has been shown that as long as traffic sources adapt their rates to aggregate congestion measure in their paths, they implicitly maximize certain utility. In this paper we study some counter-intuitive throughput behaviors in such networks, pertaining to whether a fair allocation is always inefficient and whether increasing capacity always raises aggregate throughput. A bandwidth allocation policy can be defined in terms of a class of utility functions parameterized by a scalar a that can be interpreted as a quantitative measure of fairness. An allocation is fair if /spl alpha/ is large and efficient if aggregate throughput is large. All examples in the literature suggest that a fair allocation is necessarily inefficient. We characterize exactly the tradeoff between fairness and throughput in general networks. The characterization allows us both to produce the first counter-example and trivially explain all the previous supporting examples. Surprisingly, our counter-example has the property that a fairer allocation is always more efficient. In particular it implies that maxmin fairness can achieve a higher throughput than proportional fairness. Intuitively, we might expect that increasing link capacities always raises aggregate throughput. We show that not only can throughput be reduced when some link increases its capacity, more strikingly, it can also be reduced when all links increase their capacities by the same amount. If all links increase their capacities proportionally, however, throughput will indeed increase. These examples demonstrate the intricate interactions among sources in a network setting that are missing in a single-link topology.