Abstract—We consider the problem of designing systems for the transmission of video signals of the quality found in current television broadcasts, over high-speed segments of the public IP network. Our most important contribution is the definition of a network/coder interface for IP networks which gathers channel state information, and then sets parameters of the video coder to maximize the quality of the signal delivered to the receiver, while remaining fair to other data or video connections. This interface plays a role analogous to that of a Leaky Bucket controller, in that it specifies traffic shaping parameters which result in simultaneous good Quality-of-Service (QoS) for the source and good network performance. Since the network is not assumed to provide any form of QoS guarantee, fundamental to our construction is a hidden Markov model for the channel, based on which the interface solves a problem of optimal stochastic control, to decide how to configure the encoder. Other contributions are a) modifications to the standard Internet transport protocol, to make it suitable for the transport of delay-constrained traffic and to gather channel state information, and b) the design of an error-resilient video coder. Experimental studies reveal that the proposed system is able to stream video signals of the quality of current TV-broadcasts, among hosts in wide-area networks connected to the experimental vBNS backbone. Index Terms—Communication systems, computer networks, hidden Markov models, interactive TV, protocols, stochastic systems. I.

The explosive increase in the volume and variety of Internet traffic has placed a growing emphasis on congestion control and fairness in Internet routers. Approaches to the problem of congestion, such as active queue management schemes like Random Early Detection (RED) use congestion avoidance techniques and are successful with TCP flows. Approaches to the problem of fairness, such as Fair Random Early Drop (FRED), punish misbehaved, non-TCP flows. Unfortunately, these punishment mechanisms also result in a significant performance drop for multimedia flows that use flow control. We extend ClassBased Threshold (CBT) [4], and propose a new active queue management mechanism as an extension to RED called Dynamic Class-Based Threshold (D-CBT) to improve multimedia performance on the Internet. Also, as an effort to further improve multimedia performance especially on jitter, we propose a lightweight packet scheduling called Cut-In Packet Scheduling (ChIPS) as an alternative to FIFO packet sc...

QoS Enhancement with Partial State. (August 1999) Deying Tong, B.E., Tsinghua University, China Chair of Advisory Committee: A.L.N. Reddy Considerable work has been done in devising mechanisms for providing service guarantees within a network. These schemes can be broadly classied into two categories, schemes that require maintaining state for each ow and schemes that do not require maintaining state for each ow within the network. Both the approaches have their advantages and proponents. This paper looks at a scheme, that falls in between these two extremes, where a network switch may be able to maintain state for a xed number of ows (possibly less than the number of ows it serves). This thesis looks at the services that can be provided by a limited amount of state. As a rst step, it presents SACRED, a method that employs Sampling and Caching in addition to RED at a router to enhance the QoS. The proposed mechanism uses caching to deal with the limited amount of state and us...

In this survey, we first review the concept of congestion control with a focus on the Transmission Control Protocol/Internet Protocol (TCP/IP). We describe many recently proposed algorithms to combat congestion and improve performance, particularly active queue management (AQM) algorithms such as random early detection (RED) and its variants. We then survey control-theoretic analysis and design of TCP congestion control with an AQM scheme. In addition, we discuss three problems associated with AQM proposals: parameter setting, the insensitivity to the input traffic load variation, and the mismatch between macroscopic and microscopic behavior of queue length dynamics. As alternatives to AQM algorithms, we also survey architectural approaches such as modification of source or network algorithms, and economic approaches including pricing or optimization of allocated resources. Finally, we list many open issues that persist in the design, operation, and control of the Internet. Internet congestion occurs when the aggregate demand for

Like interactive audio/video applications, distributed virtual environments (DVEs) require continuous, lowlatency delivery of media. While end-system media adaptations and network-based forwarding services have been developed to support audio/video applications, it remains an open question whether these mechanisms can be either directly applied or adapted to realize the requirements of DVEs. We present the results of a study on the use of audio/video media adaptations and router-based active queue management (AQM) to support the data-flows generated by the UNC nanoManipulator -- a DVE interface to a scanned-probe microscope. We present a delayjitter management scheme used to support a haptic forcefeedback tracking/pointing device used in the nanoManipulator and an AQM scheme based on buffer allocation in routers to reduce packet loss. The results of early experiments are promising and provide evidence that a sophisticated virtual environment interface can operate over the Internet to control a remote microscope in real-time.

Introduction & Motivation Our research at UNC concerns the management of multimedia and more general continuous-media flows across the Internet. We focus on network support for (highly) interactive applications where real-time transmission is essential for the correct operation of the application. Such applications include teleimmersion applications and interactive distributed virtual environments. In these applications minimum possible end-to-end transmission delay is required for effective device control and for maintaining the sense of immersion in a virtual world [1, 2]. To realize the performance requirements of these and other real-time multimedia applications, end-system media adaptation techniques are commonly employed to ameliorate the effects of contention for bandwidth and other resources in the network. Most multimedia and virtual environment systems, including the ones we consider, are capable of adapting to modest changes in end-to-end network delay and throughpu

Abstract — Delay sensitive applications, such as voice over IP and network games, often sacrifice throughput for lower delay to obtain better quality. Unfortunately, the Internet does not allow an application to choose the amount of delay or throughput it receives and instead packets from all applications receive the same best-effort service. This paper presents a new QoS mechanism called the Traffic Sensitive Quality of Service controller (TSQ) that provides better delay performance for delay sensitive applications and higher throughput for throughput sensitive applications. Also contributed are quality metrics for some typical Internet applications that can be used by an application to adapt its delay hints and evaluate QoS based on current Internet traffic conditions. Experiments suggest TSQ’s benefits to performance along with retention of the current best-effort Internet environment without complicated traffic monitoring or policing. I.

We consider the problem of flow coordination in distributed multimedia applications. Most transport-level protocols are designed to operate independently and lack mechanisms for sharing information with other flows and coordinating data transport in various ways. This limitation becomes problematic in distributed applications that employ numerous flows between two computing clusters sharing the same intermediary forwarding path across the Internet. In this paper, we propose an open architecture that supports the sharing of network state information, peer flow information, and application-specific information. Called simply the Coordination Protocol (CP), the scheme facilitates coordination of network resource usage across flows belonging to the same application, as well as aiding other types of coordination. The effectiveness of our approach is illustrated in the context of multi-streaming in 3D tele-immersion where consistency of network information across flows both greatly improves frame transport synchrony and minimizes buffering delay.

Current congestion control approaches that attempt to provide fair bandwidth allocation among competing flows primarily consider only data rate when making decisions on which packets to drop. However, responsive flows with high round trip times (RTTs) can still receive significantly less bandwidth than responsive flows with low round trip times. This paper proposes a congestion control scheme called WHITE that addresses router unfairness in handling flows with significantly different RTTs. Using a best-case estimate of a flow’s RTT provided in each packet by the flow source or by an edge router, WHITE computes a stabilized average RTT. The average RTT is then compared with the RTT of each incoming packet, dynamically adjusting the drop probability so as to protect the bandwidth of flows with high RTTs while curtailing the bandwidth of flows with low RTTs. We present simulation results and analysis that demonstrate that WHITE provides better fair-ness than other rate-based congestion control strategies over a wide-range of traffic conditions. The improved fairness of WHITE comes close to the fairness of Fair

The Internet, traditionally FTP, e-mail and Web traffic, is increasingly supporting emerg-ing applications such as IP telephony, video conferencing and online games. These new genres of applications have different requirements in terms of throughput and delay than traditional applications. For example, interactive multimedia applications, unlike tra-ditional applications, have more stringent delay constraints and less stringent loss con-straints. Unfortunately, the current Internet offers a monolithic best-effort service to all applications without considering their specific requirements. Adaptive RED (ARED) is an Active Queue Management (AQM) technique, which optimizes the router for through-put. Throughput optimization provides acceptable QoS for traditional throughput sensi-tive applications, but is unfair for these new delay sensitive applications. While previous work has used different classes of QoS at the router to accommodate applications with varying requirements, thus far all have provided just 2 or 3 classes of service for ap-plications to choose from. We propose two AQM mechanisms to optimize router for better overall QoS. Our first mechanism, RED-Worcester, is a simple extension to ARED