Managing QoS requirements for video streaming: from intra‐node to inter‐node

Streaming video over IP networks has become increasingly popular; however, compared to traditional data traffic, video streaming places different demands on quality of service (QoS) in a network, particularly in terms of delay, delay variation, and data loss. In response to the QoS demands of video applications, network techniques have been proposed to provide QoS within a network. Unfortunately, while efficient from a network perspective, most existing solutions have not provided end‐to‐end QoS that is satisfactory to users. In this paper, packet scheduling and end‐to‐end QoS distribution schemes are proposed to address this issue. The design and implementation of the two schemes are based on the active networking paradigm. In active networks, routers can perform user‐driven computation when forwarding packets, rather than just simple storing and forwarding packets, as in traditional networks. Both schemes thus take advantage of the capability of active networks enabling routers to adapt to the content of transmitted data and the QoS requirements of video users. In other words, packet scheduling at routers considers the correlation between video characteristics, available local resources and the resulting visual quality. The proposed QoS distribution scheme performs inter‐node adaptation, dynamically adjusting local loss constraints in response to network conditions in order to satisfy the end‐to‐end loss requirements. An active network‐based simulation shows that using QoS distribution and packet scheduling together increases the probability of meeting end‐to‐end QoS requirements of networked video. Copyright © 2005 John Wiley & Sons, Ltd.     

Providing deterministic packet delays and packet losses in multimedia wireless networks

‘Anytime, anywhere’ communication, information access and processing are much cherished in modern societies because of their ability to bring flexibility, freedom and increased efficiency to individuals and organizations. Wireless communications, by providing ubiquitous and tetherless network connectivity to mobile users, are therefore bound to play a major role in the advancement of our society. Although initial proposals and implementations of wireless communications are generally focused on near‐term voice and electronic messaging applications, it is recognized that future wireless communications will have to evolve towards supporting a wider range of applications, including voice, video, data, images and connections to wired networks. This implies that future wireless networks must provide quality‐of‐service (QoS) guarantees to various multimedia applications in a wireless environment. Typical traffic in multimedia applications can be classified as either Constant‐Bit‐Rate (CBR) traffic or Variable‐Bit‐Rate (VBR) traffic. In particular, scheduling the transmission of VBR multimedia traffic streams in a wireless environment is very challenging and is still an open problem. In general, there are two ways to guarantee the QoS of VBR multimedia streams, either deterministically or statistically. In particular, most connection admission control (CAC) algorithms and medium access control (MAC) protocols that have been proposed for multimedia wireless networks only provide statistical, or soft, QoS guarantees. In this paper, we consider deterministic QoS guarantees in multimedia wireless networks. We propose a method for constructing a packet‐dropping mechanism that is based on a mathematical framework that determines how many packets can be dropped while the required QoS can still be preserved. This is achieved by employing: (1) An accurate traffic characterization of the VBR multimedia traffic streams; (2) A traffic regulator that can provide bounded packet loss and (3) A traffic scheduler that can provide bounded packet delay. The combination of traffic characterization, regulation and scheduling can provide bounded loss and delay deterministically. This is a distinction from traditional deterministic QoS schemes in which a 0% packet loss are always assumed with deterministically bounding the delay. We performed a set of performance evaluation experiments. The results will demonstrate that our proposed QoS guarantee schemes can significantly support more connections than a system, which does not allow any loss, at the same required QoS. Moreover, from our evaluation experiments, we found that the proposed algorithms are able to out‐perform scheduling algorithms adopted in state‐of‐the‐art wireless MAC protocols, for example Mobile Access Scheme Based on Contention and Reservation for ATM (MASCARA) when the worst‐case traffic is being considered. Copyright © 2002 John Wiley & Sons, Ltd.     

A prefetching protocol for continuous media streaming in wirelessenvironments

Streaming of continuous media over wireless links is a notoriously difficult problem. This is due to the stringent quality of service (QoS) requirements of continuous media and the unreliability of wireless links. We develop a streaming protocol for the real-time delivery of prerecorded continuous media from (to) a central base station to (from) multiple wireless clients within a wireless cell. Our protocol prefetches parts of the ongoing continuous media streams into prefetch buffers in the clients (base station). Our protocol prefetches according to a join-the-shortest-queue (JSQ) policy. By exploiting rate adaptation techniques of wireless data packet protocols, the JSQ policy dynamically allocates more transmission capacity to streams with small prefetched reserves. Our protocol uses channel probing to handle the location-dependent, time-varying, and bursty errors of wireless links. We evaluate our prefetching protocol through extensive simulations with VBR MPEG and H.263 encoded video traces. Our simulations indicate that for bursty VBR video with an average rate of 64 kb/s and typical wireless communication conditions our prefetching protocol achieves client starvation probabilities on the order of 10^-4 and a bandwidth efficiency of 90% with prefetch buffers of 128 kbytes     

改善Internet视频流QoS的途径和方法

在当前Internet的尽力而为的服务模式下，网络拥塞和分组丢失不可避免，视频流必须使用有效的拥塞控制和差错控制来改善性能。本文分析了：Internet视频流QoS影响因素，提出了两种QoS解决方案：基于终端和基于网络。本文着重讨论了基于终端的QoS解决方案，在目前Internet的环境下，基于终端的QoS解决方案更具可行性。     

Design of a fair bandwidth allocation policy for VBR traffic in ATMnetworks

Since real-time variable bit rate (VBR) traffic is inherently bursty, dynamic bandwidth allocation is necessary for ATM streams that carry VBR traffic. In order to provide quality-of-services (QoS) guarantees and to reduce the computational complexity, an hybrid of guaranteed and dynamic adaptive allocation scheme requires to be implemented. Typical dynamic allocations to competing streams are done in the form of linear proportions to the bandwidth requirements. We show that during temporary link congestion such proportional arrangements can give rise to unequal queue growth and, subsequently, degraded QoS. This is found to be true even for streams that belong to the same VBR class and share identical long term traffic characteristics and QoS requirements. In this paper, four allocation algorithms are presented and analyzed in terms of their fairness and QoS potential for real-time VBR traffic. We propose and show that a novel allocation strategy, termed Minmax, solves the mentioned problem of unfairness within a class. By maintaining a fair distribution of buffer length across the streams of a class, the proposed policy can achieve better and fairer QoS performance compared to the traditional methods. We present analytical results, proofs and a simulation study of the described algorithms. Four allocation policies for handling MPEG VBR video streams are simulated in the context of a wireless ATM (WATM) medium access control. The results show that in certain scenarios, the Minmax strategy can reduce losses by an order of magnitude, while decreasing delays substantially     

Efficient Traffic Scheduling for Real Time VBR MPEG Video Transmission Over DOCSIS-Based HFC Networks

Data-over-cable service interface specifications (DOCSIS), the de facto standard in the cable industry, defines a scheduling service called real-time polling service (rtPS) to provision quality of service (QoS) transmission of real-time variable bit rate (VBR) videos. However, the rtPS service intrinsically has high latency, which makes it not applicable to real-time traffic transport. In this paper, we present a novel traffic scheduling algorithm for hybrid fiber coax (HFC) networks based on DOCSIS that aims to provide QoS for real-time VBR video transmissions. The novel characteristics of this algorithm, as compared to those described in published literatures, include 1) it predicts the bandwidth requirements for future traffic using a novel traffic predictor designed to provide simple yet accurate online prediction; and 2) it takes the attributes of physical (PHY) layer, media access control (MAC) layer and application layer into consideration. In addition, the proposed traffic scheduling algorithm is completely compatible with the DOCSIS specification and does not require any protocol changes. We analyze the performance of the proposed traffic predictor and traffic scheduling algorithm using real-life MPEG video traces. Simulation results indicate that 1) the proposed traffic predictor significantly outperforms previously published techniques with respect to the prediction error and 2) Compared with several existing scheduling algorithms, the proposed traffic scheduling algorithm surpasses other mechanisms in terms of channel utilization, buffer usage, packet delay, and packet loss rate.     

Leaky-bucket based rate control algorithm for C VBR video encoding

A rate control algorithm is presented for constrained variable bit rate (C VBR) encoding to make the compressed video stream more friendly to the user network interface (UNI) than in free VBR coding. Experiments show that the algorithm would not only meet the delay constraint and traffic contract of VBR transfer so as to avoid the cell loss over UNI, but also benefit the statistical multiplexing by reducing the burstiness of the MPEG encoded stream     

An Approach to Scheduling Bursty Traffic

The scheduling scheme in packet switching networks is one of the most critical features that can affect the performance of the network. Hence, many scheduling algorithms have been suggested and some indices, such as fairness and latency, have been proposed for the comparison of their performances. While the nature of Internet traffic is bursty, traditional scheduling algorithms try to smooth the traffic and serve the users based on this smoothed traffic. As a result, the fairness index mainly considers this smoothed traffic and the service rate as the main parameter to differentiate among different sessions or flows. This work uses burstiness as a differentiating factor to evaluate scheduling algorithms proposed in the literature. To achieve this goal, a new index that evaluates the performance of a scheduler with bursty traffic is introduced. Additionally, this paper introduces a new scheduler that not only uses arrival rates but also considers burstiness parameters in its scheduling algorithms.     

On the effects of traffic mixtures in direct broadcast satellite networks

In this paper we present a multi‐criterion control simulation in a realistically complex environment of a satellite network, involving non‐symmetric up and downlinks. Direct broadcast satellite (DBS) networks carrying heterogeneous traffic is characterized with challenges, such as high traffic burstiness, wireless channel dynamics, and large, but limited capacity. On the other hand, there are system characteristics that can be leveraged to address these challenges such as in centralized topology, different levels in quality of service (QoS) and priorities, availability of side information about channel conditions, flexibility in delivery of delay insensitive traffic, etc. We have developed an adaptive resource allocation and management (ARAM) system that takes the advantage of such characteristics to maximize the utilization of the available capacity on the forward DBS link, while maintaining QoS in the presence of channel effects and congestion in the network. Since variable‐bit‐rate (VBR) video traffic is given priority over available‐bit‐rate (ABR) data traffic in the ARAM concept, in this paper we investigate the impact of the fraction of VBR load in overall load. Copyright © 2002 John Wiley & Sons, Ltd.     

Towards the formation of comprehensive SLAs between heterogeneous wireless DiffServ domains

Traffic patterns generated by multimedia services are different from traditional Poisson traffic. It has been shown in numerous studies that multimedia network traffic exhibits self-similarity and burstiness over a large range of time-scales. The area of wireless IP traffic modeling for the purpose of providing assured QoS to the end-user is still immature and the majority of existing work is based on characterization of wireless IP traffic without any coupling of the behaviour of queueing systems under such traffic conditions. Work in this area has either been limited to simplified models of FIFO queueing systems which do not accurately reflect likely queueing system implementations or the results have been limited to simplified numerical analysis studies. In this paper, we advance the knowledge of queueing systems by example of traffic engineering of different UMTS service classes. Specifically, we examine QoS mapping using three common queueing disciplines; Priority Queuing (PQ), Low Latency Queuing (LLQ) and Custom Queueing (CQ), which are likely to be used in future all-IP based packet transport networks. The present study is based on a long-range dependent traffic model, which is second order self-similar. We consider three different classes of self-similar traffic fed into a G/M/1 queueing system and construct analytical models on the basis of non-preemptive priority, low-latency queueing and custom queueing respectively. In each case, expressions are derived for the expected waiting times and packet loss rates of different traffic classes. We have developed a comprehensive discrete-event simulator for a G/M/1 queueing system in order to understand and evaluate the QoS behaviour of self-similar traffic and carried out performance evaluations of multiple classes of input traffic in terms of expected queue length, packet delay and packet loss rate. Furthermore, we have developed a traffic generator based on the self-similar traffic model and fed the generated traffic through a CISCO router-based test bed. The results obtained from the three different queueing schemes (PQ, CQ and LLQ) are then compared with the simulation results in order to validate our analytical models.     

Efficiency of the GSM-GPRS Air Interface for Real-Time IP Traffic Flows With and Without Packet Dropping

One IP terminal can occupy a single slot or a multiple number of slots within time frames in the GSM and GPRS, respectively. A limited number of radio resources (slots) are allocated in a base station for such IP terminals. If one IP terminal can occupy only one slot discontinuously in a time frame, there is one possibility resorting to all IP terminals to preserve active mode at a time. Thus, the number of accepted call in the GSM is the same as that of the radio resource. Similarly, if one terminal can occupy a multiple number of slots discontinuously/dynamically in a time frame, the number of accepted calls is obtained by dividing the number of radio resources during that time by the maximum allowed number of slots per IP terminal. A burstiness factor is defined for the IP traffic over GSM-GPRS air interface. Traffic channel efficiency with a bursty real-time IP traffic is unacceptably low, especially with the range of acceptable call loss probabilities pertaining to a lower burstiness factor. The channel efficiency can be enhanced and the call loss probability can be suppressed significantly if a higher maximum number of calls is accepted. Allocated radio resources are less than the maximum number of packet transmissions at a time. Therefore, some packets could be dropped from the real-time transmission system. A complete analysis for the real-time IP packet transmission over the single slot GSM and dynamically variable multislot GPRS air interface without packet dropping, and with packet dropping that increases the channel efficiency is executed. Results show that the channel efficiency as well as the packet dropping probability increases with increasing call intensity, maximum number of admitted IP calls and the burstiness factor.     

Challenges and opportunities of delivering IP-based residential television services

This article describes the main challenges of implementing an end-to-end architecture for delivery of high-quality, IP-based residential TV services to residential customers. The IP-based approach can be implemented with an IP multicast overlay network with traditional routers or use IP-multicast-aware ATM switching systems. Both approaches use IP multicast to transport MPEG-2 broadcast video and can work on any access architecture, especially on copper-based architectures (xDSL) such as ASDL and VDSL. The main challenges met while implementing the IP-based architecture are competitive positioning relative to traditional CATV architectures, overall architecture, head-end architecture and quality issues, traffic engineering for stringent QoS requirements, IP multicast requirements, and business case considerations. The IP-based approach described leverages Internet technology advancements and capitalizes on the impacts of Internet on interactive entertainment. Video channel manipulation at the head-end is dependent on access bandwidth and affects video quality. Video traffic management to meet stringent QoS requirements is challenging at the IP layer. High-capacity, responsive IP multicasting is essential to achieving high service quality. Cost-effective IP multicasting is a critical component of the business case.     

The transmission of MPEG‐2 VBR video under usage parameter control

This paper studies the transmission of MPEG‐2 VBR video over ATM network under usage parameter control. The idea is to seek a compromise between the network utilization and the quality of video service by applying UPC‐based rate control strategies to the video source. A modified leaky bucket algorithm is proposed to calculate the constraints on the bit‐rate guaranteeing conformance to peak cell rate, sustainable cell rate and burst tolerance usage parameters. Two rate control strategies, one for real‐time generated video coding and the other for pre‐compressed video, are proposed for MPEG‐2 VBR video. The rate control strategies control the video source to generate traffic conforming to the constraints on the bit rate. The experimental results show that both the UPC‐based rate control strategies can provide lossless transmission from the source perspective as well as to reduce the burstiness of the traffic. To keep within the bit‐rate allowed, the control method uses coarser quantization to maintain better picture quality than that by removing the number of AC transformed coefficients. The slight degradation of picture quality caused by the source rate control is preferable than the severe drop of picture quality caused by the cell loss at UPC. Copyright © 2001 John Wiley & Sons, Ltd.     

Mobile flow-aware networks for mobility and QoS support in the IP-based wireless networks

As the volume of mobile traffic consisting of video, voice, and data is rapidly expanding, a challenge remains with the mobile transport network, which must deliver data traffic to mobile devices without degrading the service quality. Since every Internet service holds its own service quality requirements, the flow-aware traffic management in fine granularity has been widely investigated to guarantee Quality of Service (QoS) in the IP networks. However, the mobile flow-aware management has not been sufficiently developed yet because of the inherent constraints of flow routing in the mobile networks regarding flow-aware mobility and QoS support. In this paper, we propose a flow-aware mobility and QoS support scheme called mobile flow-aware network (MFAN) for IP-based wireless mobile networks. The proposed scheme consists of dynamic handoff mechanisms based on QoS requirements per flow to reduce the processing overhead of the flow router while ensuring QoS guarantee to mobile flows. The performance analyses of the proposed scheme demonstrate that MFAN successfully supports the mobile flow traffic delivery while satisfying the QoS requirement of flows in the wireless mobile IP networks.     

End-to-End Adaptive QoS Provisioning over GPRS Wireless Mobile Network

The General Packet Radio Service (GPRS) offers performance guaranteed packet data services to mobile users over wireless frequency-division duplex links with time division multiple access, and core packet data networks. This paper presents a dynamic adaptive guaranteed Quality-of-Service (QoS) provisioning scheme over GPRS wireless mobile links by proposing a guaranteed QoS media access control (GQ-MAC) protocol and an accompanying adaptive prioritized-handoff call admission control (AP-CAC) protocol to maintain GPRS QoS guarantees under the effect of mobile handoffs. The GQ-MAC protocol supports bounded channel access delay for delay-sensitive traffic, bounded packet loss probability for loss-sensitive traffic, and dynamic adaptive resource allocation for bursty traffic with peak bandwidth allocation adapted to the current queue length. The AP-CAC protocol provides dynamic adaptive prioritized admission by differentiating handoff requests with higher admission priorities over new calls via a dynamic multiple guard channels scheme, which dynamically adapts the capacity reserved for dealing with handoff requests based on the current traffic conditions in the neighboring radio cells. Integrated services (IntServ) QoS provisioning over the IP/ATM-based GPRS core network is realized over a multi-protocol label switching (MPLS) architecture, and mobility is supported over the core network via a novel mobile label-switching tree (MLST) architecture. End-to-end QoS provisioning over the GPRS wireless mobile network is realized by mapping between the IntServ and GPRS QoS requirements, and by extending the AP-CAC protocol from the wireless medium to the core network to provide a unified end-to-end admission control with dynamic adaptive admission priorities.     

LÉvy Flights and Fractal Modeling of Internet Traffic

The relation between burstiness and self-similarity of network traffic was identified in numerous papers in the past decade. These papers suggested that the widely used Poisson based models were not suitable for modeling bursty, local-area and wide-area network traffic. Poisson models were abandoned as unrealistic and simplistic characterizations of network traffic. Recent papers have challenged the accuracy of these results in today's networks. Authors of these papers believe that it is time to reexamine the Poisson traffic assumption. The explanation is that as the amount of Internet traffic grows dramatically, any irregularity of the network traffic, such as burstiness, might cancel out because of the huge number of different multiplexed flows. Some of these results are based on analyses of particular OC48 Internet backbone connections and other historical traffic traces. We analyzed the same traffic traces and applied new methods to characterize them in terms of packet interarrival times and packet lengths. The major contribution of the paper is the application of two new analytical methods. We apply the theory of smoothly truncated Levy flights and the linear fractal model in examining the variability of Internet traffic from self-similar to Poisson. The paper demonstrates that the series of interarrival times is still close to a self-similar process, but the burstiness of the packet lengths decreases significantly compared to earlier traces.      

A Model‐Based Analysis of Secure Video Transmission Based on IPSec and IPv4

A promising solution to protect wired Internet networks is to use the Secure Internet Protocol (IPSec); however, this has some drawbacks, particularly on the quality of service (QoS). This paper aims at evaluating the video traffic QoS in terms of end‐to‐end delay and packet loss rate. Based on some basic assumptions, our analysis shows that the performance with IPSec is rapidly inferior to the IPv4 performance. We thus suggest adding some QoS parameters into IPSec in order to achieve a compromise between QoS and security.     

ATM in MPLS-based converged core data networks

Services supported by asynchronous transfer mode account for the majority of data and Internet service revenues generated by carrier networks today. This is based on ATM's ability to support high availability services with quality of service. However, the influences of the Internet and a highly dynamic telecommunications market have raised demands for increased flexibility while controlling costs. Therefore, future carrier networks are likely to continue to be based on established technologies, such as ATM, as well as IP. In many cases, this is achieved through maintaining separate ATM and IP core networks, with the IP network supporting Internet services, while the ATM network continues to support guaranteed services such as private lines, broadband access, and video. In some cases, however, it can be advantageous for a carrier to transport segments of their ATM network over their IP network core; for example, to transport ATM traffic currently carried on leased facilities onto an IP network where the service provider owns the facilities. Developments in IP and MPLS-based traffic engineering and QoS may increase the ability of IP-based networks to support ATM services using MPLS. This article provides an overview of approaches enabling a network based on MPLS that naturally supports IP services to also support ATM services. The drivers and requirements for convergence on an IP/MPLS core network are presented, followed by an overview of the different approaches and associated challenges currently being debated in the standards bodies.     

Performance of delay‐sensitive traffic in multi‐layered satellite IP networks with on‐board processing capability

In this article, performance of delay‐sensitive traffic in multi‐layered satellite Internet Protocol (IP) networks with on‐board processing (OBP) capability is investigated. With OBP, a satellite can process the received data, and according to the nature of application, it can decide on the transmission properties. First, we present a concise overview of relevant aspects of satellite networks to delay‐sensitive traffic and routing. Then, in order to improve the system performance for delay‐sensitive traffic, specifically Voice over Internet Protocol (VoIP), a novel adaptive routing mechanism in two‐layered satellite network considering the network's real‐time information is introduced and evaluated. Adaptive Routing Protocol for Quality of Service (ARPQ) utilizes OBP and avoids congestion by distributing traffic load between medium‐Earth orbit and low‐Earth orbit layers. We utilize a prioritized queueing policy to satisfy quality‐of‐service (QoS) requirements of delay‐sensitive applications while evading non‐real‐time traffic suffer low performance level. The simulation results verify that multi‐layered satellite networks with OBP capabilities and QoS mechanisms are essential for feasibility of packet‐based high‐quality delay‐sensitive services which are expected to be the vital components of next‐generation communications networks. Copyright © 2007 John Wiley & Sons, Ltd.