Distributed Rate Allocation Policies for Multi-Homed Video Streaming over Heterogeneous Access Networks

We consider the problem of rate allocation among multiple simultaneous video streams sharing multiple heterogeneous access networks. We develop and evaluate an analytical framework for optimal rate allocation based on observed available bit rate (ABR) and round-trip time (RTT) over each access network and video distortion-rate (DR) characteristics. The rate allocation is formulated as a convex optimization problem that minimizes the total expected distortion of all video streams. We present a distributed approximation of its solution and compare its performance against H-infinity optimal control and two heuristic schemes based on TCP-style additive-increase-multiplicative decrease (AIMD) principles. The various rate allocation schemes are evaluated in simulations of multiple high-definition (HD) video streams sharing multiple access networks. Our results demonstrate that, in comparison with heuristic AIMD-based schemes, both media-aware allocation and H-infinity optimal control benefit from proactive congestion avoidance and reduce the average packet loss rate from 45% to below 2%. Improvement in average received video quality ranges between 1.5 to 10.7 dB in PSNR for various background traffic loads and video playout deadlines. Media-aware allocation further exploits its knowledge of the video DR characteristics to achieve a more balanced video quality among all streams.Comment: 12 pages, 22 figure

SRC: Stable Rate Control for Streaming Media

Rate control, in conjunction with congestion control, is important and necessary to maintain both stability of overall network and high quality of individual data transfer flows. In this paper, we study stable rate control algorithms for streaming data, based on control theory. We introduce various control rules to maintain both sending rate and receiver buffer stable. We also propose an adaptive two-state control mechanism to ensure the rate control algorithms are compatible to TCP traffics. Extensive experimental results are shown to demonstrate the effectiveness of the rate control algorithms

Combined use of congestion control and frame discarding for Internet video streaming

Cataloged from PDF version of article.Increasing demand for video applications over the Internet and the inherent uncooperative behavior of the User Datagram Protocol (UDP) used currently as the transport protocol of choice for video networking applications, is known to be leading to congestion collapse of the Internet. The congestion collapse can be prevented by using mechanisms in networks that penalize uncooperative flows like UDP or employing end-to-end congestion control. Since today’s vision for the Internet architecture is based on moving the complexity towards the edges of the networks, employing end-to-end congestion control for video applications has recently been a hot area of research. One alternative is to use a Transmission Control Protocol (TCP)-friendly end-to-end congestion control scheme. Such schemes, similar to TCP, probe the network for estimating the bandwidth available to the session they belong to. The average bandwidth available to a session using a TCP-friendly congestion control scheme has to be the same as that of a session using TCP. Some TCP-friendly congestion control schemes are highly responsive as TCP itself leading to undesired oscillations in the estimated bandwidth and thus fluctuating quality. Slowly responsive TCP-friendly congestion control schemes to prevent this type of behavior have recently been proposed in the literature. The main goal of this thesis is to develop an architecture for video streaming in IP networks using slowly responding TCP-friendly end-to-end congestion control. In particular, we use Binomial Congestion Control (BCC). In this architecture, the video streaming device intelligently discards some of the video packets of lesser priority before injecting them in the network in order to match the incoming video rate to the estimated bandwidth using BCC and to ensure a high throughput for those video packets with higher priority. We iiidemonstrate the efficacy of this architecture using simulations in a variety of scenarios.Yücesan, OngunM.S

Dual-Mode Congestion Control Mechanism for Video Services

Recent studies have shown that video services represent over half of Internet traffic, with a growing trend. Therefore, video traffic plays a major role in network congestion. Currently on the Internet, congestion control is mainly implemented through overprovisioning and TCP congestion control. Although some video services use TCP to implement their transport services in a manner that actually works, TCP is not an ideal protocol for use by all video applications. For example, UDP is often considered to be more suitable for use by real-time video applications. Unfortunately, UDP does not implement congestion control. Therefore, these UDP-based video services operate without any kind of congestion control support unless congestion control is implemented on the application layer. There are also arguments against massive overprovisioning. Due to these factors, there is still a need to equip video services with proper congestion control.Most of the congestion control mechanisms developed for the use of video services can only offer either low priority or TCP-friendly real-time services. There is no single congestion control mechanism currently that is suitable and can be widely used for all kinds of video services. This thesis provides a study in which a new dual-mode congestion control mechanism is proposed. This mechanism can offer congestion control services for both service types. The mechanism includes two modes, a backward-loading mode and a real-time mode. The backward-loading mode works like a low-priority service where the bandwidth is given away to other connections once the load level of a network is high enough. In contrast, the real-time mode always demands its fair share of the bandwidth.The behavior of the new mechanism and its friendliness toward itself, and the TCP protocol, have been investigated by means of simulations and real network tests. It was found that this kind of congestion control approach could be suitable for video services. The new mechanism worked acceptably. In particular, the mechanism behaved toward itself in a very friendly way in most cases. The averaged TCP fairness was at a good level. In the worst cases, the faster connections received about 1.6 times as much bandwidth as the slower connections

Adaptive filtering of MPEG system streams in IP networks

Congestion and large differences in available link bandwidth create challenges for the design of applications that want to deliver high quality video over the Internet. We present an efficient adaptive filter for MPEG System streams that can be placed in the network (e.g., as an active service). This filter adjusts the bandwidth demands of an MPEG System stream to the available bandwidth without transcoding while maintaining synchronization between the streams embedded in the MPEG System. The filter is network-friendly: it is fair with respect to other (TCP) competing streams and it avoids generating bursty traffic. This paper presents the system architecture and an evaluation of our implementation in three different operating environments: a networking testbed in a laboratory environment, a home-user scenario (DSL line with 640Kbit/s), and a wide area network covering the Atlantic (server in Europe, client in the US). Moreover we examine the network-friendliness of the adaptation protocol and the relationship between the quality of the received continuous media and the protocol's aggressiveness. Our architecture is based on efficient MPEG System filtering to achieve high-quality video over best-effort network

Scalable reliable on-demand media streaming protocols

This thesis considers the problem of delivering streaming media, on-demand, to potentially large numbers of concurrent clients. The problem has motivated the development in prior work of scalable protocols based on multicast or broadcast. However, previous protocols do not allow clients to efficiently: 1) recover from packet loss; 2) share bandwidth fairly with competing flows; or 3) maximize the playback quality at the client for any given client reception rate characteristics. In this work, new protocols, namely Reliable Periodic Broadcast (RPB) and Reliable Bandwidth Skimming (RBS), are developed that efficiently recover from packet loss and achieve close to the best possible server bandwidth scalability for a given set of client characteristics. To share bandwidth fairly with competing traffic such as TCP, these protocols can employ the Vegas Multicast Rate Control (VMRC) protocol proposed in this work. The VMRC protocol exhibits TCP Vegas-like behavior. In comparison to prior rate control protocols, VMRC provides less oscillatory reception rates to clients, and operates without inducing packet loss when the bottleneck link is lightly loaded. The VMRC protocol incorporates a new technique for dynamically adjusting the TCP Vegas threshold parameters based on measured characteristics of the network. This technique implements fair sharing of network resources with other types of competing flows, including widely deployed versions of TCP such as TCP Reno. This fair sharing is not possible with the previously defined static Vegas threshold parameters. The RPB protocol is extended to efficiently support quality adaptation. The Optimized Heterogeneous Periodic Broadcast (HPB) is designed to support a range of client reception rates and efficiently support static quality adaptation by allowing clients to work-ahead before beginning playback to receive a media file of the desired quality. A dynamic quality adaptation technique is developed and evaluated which allows clients to achieve more uniform playback quality given time-varying client reception rates