Evolution of the Internet QoS and support for soft real-time applications

The past few years have witnessed the emergence of many real-time networked applications on the Internet. These types of applications require special support from the underlying network such as reliability, timeliness, and guaranteed delivery, as well as different levels of service quality. Unfortunately, this support is not available within the current "best-effort" Internet architecture. In this paper, we review several mechanisms and frameworks proposed to provide network- and application-level quality of service (QoS) in the next-generation Internet. We first discuss the QoS requirements of many of the above-mentioned real-time applications, and then we categorize them according to the required service levels. We also describe the various building blocks often used in QoS approaches. We briefly present asynchronous transfer mode (ATM) and Internet Protocol precedence. Then, we present and compare two service architectures recently adopted by the Internet Engineering Task Force, called integrated services (IntServ) and differentiated services (DiffServ), for providing per-flow and aggregated-flow service guarantees, respectively. We focus on DiffServ because it is a candidate QoS framework to be used in next-generation Internet along with multiprotocol label switching and traffic engineering. We also examine several operational and research issues that need to be resolved before such frameworks can be put in practice.     

Supporting IP QoS in the General Packet Radio Service

The General Packet Radio Service is the current enhancement in the GSM infrastructure, capable of handling Internet protocol traffic for mobile computing and communications. A major deficiency of the current GPRS specification is the lack of adequate IP quality of service support. Two schemes for enhancing the GPRS architecture with the existing IP QoS support architectures, IntServ and DiffServ, are proposed. Solutions are proposed to the problem of establishing QoS reservations across the GPRS core network, and the required signaling enhancements and modifications in the components of the GPRS architecture are identified. Of the two proposed schemes the IntServ one requires frequent refreshing of state information and extra signaling. To quantify the effect that signaling overhead has on GPRS operation and performance, a simulation model of the proposed IntServ architecture was developed, which includes models of the GPRS cellular infrastructure, network traffic, and user movement. The obtained simulation results show that the proposed IntServ architecture demonstrated good scalability, even for large user populations     

A novel analytical framework compounding statistical traffic modeling and aggregate-level service curve disciplines: network performance and efficiency implications

This paper demonstrates that higher network resource efficiency can be achieved by using resource management protocols which consider service disciplines based on service curves together with statistical traffic modeling. To this end, an appropriate analytical framework is introduced which allows calculation of the performance statistically guaranteed to any flow out of an aggregate. This feature enables the analytical framework to be applied to the elements of the core network where aggregates of traffic are considered instead of single flows in order to avoid scalability problems. Given that flows are modeled in the analytical framework through switched batch Bernoulli processes (SBBPs), the whole queueing system is denoted as SBBP/Sc/1/K. The performance is calculated in terms of loss probability and delay distribution. The proposed framework is applied in a significant multinode case study.     

Internet services beyond best effort

The packet-based data transmission of the Internet allows the multiplexing of a variety of simultaneous connections originating from sources with different characteristics (e.g. voice, video, data). Traditional Internet is based on a best effort service, whereas future Internet services make more and more special demands on the communication network. This paper presents a general overview of fundamental mechanisms for guaranteeing Quality of Service (QoS) in the current and future Internet. After introducing general QoS concepts, basic mechanisms in network routers are investigated. Furthermore, the two basic architectures of Integrated Services (IntServ) and Differentiated Services (DiffServ) are discussed in detail. Finally, the analytical investigation of a special Voice-over-IP scenario demonstrates the applicability of the relevant QoS concepts and their positive consequences with respect to the quality of this voice service. A short overview of current hot research topics concludes the paper.     

Flow‐Based QoS Management Architectures for the Next Generation Network

At the extremes of the complexity‐performance plane, there are two exemplary QoS management architectures: Integrated Services (IntServ) and Differentiated Services (DiffServ). IntServ performs ideally but is not scalable. DiffServ is simple enough to be adopted in today's core networks, but without any performance guarantee. Many compromise solutions have been proposed. These schemes, called quasi‐stateful IntServ or stateful DiffServ, however, have not attracted much attention due to their inherently compromising natures. Two disruptive flow‐based architectures have been recently introduced: the flow‐aware network (FAN) and the flow‐state‐aware network (FSA). FAN's control is implicit without any signaling. FSA's control is even more sophisticated than that of IntServ. In this paper, we survey established QoS architectures, review disruptive architectures, discuss their rationales, and points out their disadvantages. A new QoS management architecture, flow‐aggregate‐based services (FAbS), is then proposed. The FAbS architecture has two novel building blocks: inter‐domain flow aggregation and endpoint implicit admission control.     

Switched bath Bernoulli process (SBBP) and the discrete-timeSBBP/G/1 queue with application to statistical multiplexer performance

A doubly stochastic point process is proposed and analyzed. It is a switched batch Bernoulli process (SBBP) for modeling bursty and correlated input to discrete-time queuing systems. Through the investigation of the counting process, statistical characterization measures of the SBBP are explicitly obtained. A discrete-time single-server queue with SBBP input and general service time (SBBP/G/1) is then considered. The SBBP/G/1 queue has a potential applicability to a statistical multiplexer in the network. The supplementary variable technique is used to obtain the probability generating functions of performance measures in the SBBP/G/1 queue. It is shown how the SBBP is suitable for analyzing a discrete-time queue with bursty and correlated input     

A Comparison of the Utilization Efficiency between a Stateful and a Stateless Admission Control in IP Networks in a Heterogeneous Traffic Case

The IETF developed two main approaches to provide QoS aware services in the Internet: Integrated Services (IntServ) and Differentiated Services (DiffServ). Both have well known pros and cons (e.g., [Huston, 24; Bernet et al., 2]). The stateful IntServ has a greater level of accuracy and a finer level of granularity. The stateless DiffServ possesses excellent scaling properties, but lacks a standardized admission control scheme and, upon overload in a given service class, degradation of service can occur. To provide QoS in DiffServ, three possible strategies are: (i) plain and heavy over-dimensioning; (ii) admission control at the borders of the DiffServ region, coupled with suitable assumptions on the distribution of the traffic within the region, which can lead to over-dimensioning, even if less severe than the previous one; (iii) per-node admission control within the region. Following RFC2990, we recently proposed an “admission control function which can determine whether to admit a service differentiated flow along the nominated network path” [Huston, 24], i.e., the third of the above strategies. This function, named GRIP (Gauge and Gate Reservation with Independent Probing), can provide strict QoS guarantees by means of stateless DiffServ-compliant procedures. This feature is paid with a potential loss of efficiency, with respect to an ideal, stateful admission control. The goal of this paper is to evaluate analytically such loss of efficiency, in a specific heterogeneous scenario. In other words, we want to estimate how much resources we can waste if we go stateless and avoid state maintenance functions. The comparison between stateless and stateful approaches is performed under the constraint of strictly offering the same performance levels, in terms of, e.g., loss probability and delay.     

Flow‐Based Admission Control Algorithm in the DiffServ‐Aware ATM‐Based MPLS Network

This paper proposes a flow‐based admission control algorithm through an Asynchronous Transfer Mode (ATM) based Multi‐Protocol Label Switching (MPLS) network for multiple service class environments of Integrated Service (IntServ) and Differentiated Service (DiffServ). We propose the Integrated Packet Scheduler to accommodate IntServ and Best Effort traffic through the DiffServ‐aware MPLS core network. The numerical results of the proposed algorithm achieve reliable delay‐bounded Quality of Service (QoS) performance and reduce the blocking probability of high priority service in the DiffServ model. We show the performance behaviors of IntServ traffic negotiated by end users when their packets are delivered through the DiffServ‐aware MPLS core network. We also show that ATM shortcut connections are well tuned with guaranteed QoS service. We validate the proposed method by numerical analysis of its performance in such areas as throughput, end‐to‐end delay and path utilization.     

Performance analysis of an ATM multiplexer loaded with VBR trafficgenerated by multimode speech coders

Multimode coders are able to exploit the different characteristics of the speech waveform and to take into account the different peculiarities of background noise, thus allowing improvements in both signal reconstruction and network-offered load. In this context the variable rate code excited linear prediction (VR-CELP) coding, that is, a multimode variable bit rate (VBR) coding based on the CELP technique, has been introduced in the literature and is currently being considered for use in various applications, especially in the third-generation UMTS cellular systems. The target of the paper is to introduce an efficient and accurate framework allowing a network designer to analyze the impact of multimode VBR speech coding on the quality of service (QoS) provided by a wireless/wired ATM network. In order to capture the coder output characteristics, we propose to model a VR-CELP voice source by using a switched batch Bernoulli process (SBBP). More specifically, three models are introduced and compared in terms of accuracy and simplicity in determining network performance. As a result of the comparison, a four-state model has been chosen as the best tradeoff. The model is then used to analytically derive the loss probability and the jitter probability density function of an ATM multiplexer loaded by a number of VR-CELP sources. Finally, the proposed paradigm has been assessed in a case study where we demonstrate that, for a given output ATM link capacity and for a number of telecommunication services involving voice transmission, VR-CELP coding performs better than traditional on-off coding     

A discrete-time paradigm to evaluate skew performance in amultimedia ATM multiplexer

One of the most important quality-of-service parameters in a multimedia environment is skew, defined as the difference between the delays suffered by the monomedia flows belonging to the same multimedia stream. An analytical paradigm is proposed to evaluate the skew affecting a multimedia traffic stream in an asynchronous transfer mode multiplexer. For this purpose, the emission process of each multimedia source loading the multiplexer is defined as the superposition of heterogeneous correlated emission processes, each of which models one monomedia source as a switched batch Bernoulli process (SBBP). In order to model the intermedia relationships, the transition probabilities in the Markov chain underlying each SBBP are functions of the state of the other monomedia sources. The model is applied to a case study and the dependence of skew performance on some of the source characteristics, such as intermedia correlation, and some of the environment characteristics, such as buffer size, output capacity, and buffer utilization is analyzed and discussed     

Software‐defined network‐based prioritization to avoid video freezes in HTTP adaptive streaming

HTTP adaptive streaming (HAS) is becoming the de facto standard for video streaming services over the Internet. In HAS, each video is segmented and stored in different qualities. Rate adaptation heuristics, deployed at the client, allow the most appropriate quality level to be dynamically requested, based on the current network conditions. It has been shown that state‐of‐the‐art heuristics perform suboptimal when sudden bandwidth drops occur, therefore leading to freezes in the video playout, the main factor influencing users' quality of experience (QoE). This issue is aggravated in case of live events, where the client‐side buffer has to be kept as small as possible in order to reduce the playout delay between the user and the live signal. In this article, we propose a framework capable of increasing the QoE of HAS clients by reducing video freezes. The framework is based on OpenFlow, a widely adopted protocol to implement the software‐defined networking principle. An OpenFlow controller is in charge of introducing prioritized delivery of HAS segments, based on the network conditions and the HAS clients' status. Particularly, the HAS clients' status is obtained without any explicit clients‐to‐controller communication, and thus, no extra signaling is introduced into the network. Moreover, this OpenFlow controller is transparent to the quality decision process of the clients, as it assists the delivery of the segments, but it does not determine the actual quality to be requested. In order to provide a comprehensive analysis of the proposed approach, we investigate the performance of the proposed OpenFlow‐based framework in the presence of realistic Internet cross‐traffic. Particularly, we model two types of applications, namely, HTTP web browsing and progressive download video streaming, which currently represent the majority of Internet traffic together with HAS. By evaluating this novel approach through emulation in several multi‐client scenarios, we show how the proposed approach can reduce freeze time for the HAS clients due to network congestion up to 10 times compared with state‐of‐the‐art heuristics, without impacting the performance of the cross‐traffic applications. Copyright © 2016 John Wiley & Sons, Ltd.     

A Novel Analytical Framework for Integrated Cross-Layer Study of Call-Level and Packet-Level QoS in Wireless Mobile Multimedia Networks

We present a novel integrated analytical framework for analyzing the quality-of-service (QoS) performance measures in a wireless mobile multimedia network. The framework integrates physical, radio link, and network layer parameters and protocols to analyze the call-level and packet-level performances. In the network layer, call admission control (CAC) is responsible for deciding whether an incoming call can be accepted or not so that the performances of the ongoing calls do not deteriorate below the acceptable level. Also, an adaptive channel allocation (ACA) scheme is used to maximize the utilization of the radio resources. In the data link layer, queue management and error control are used for non-real-time loss-sensitive traffic. In the physical layer, a finite state Markov channel (FSMC) is used to model channel fading, and adaptive modulation is used for rate adaptation according to channel quality. Various call-level and packet-level QoS measures for real-time, non-real-time, and best-effort traffic are obtained. The analytical results are validated by extensive simulations. Examples of the applications of the presented analytical framework are also provided     

Large-Scale Network Parameter Configuration Using an On-Line Simulation Framework

As the Internet infrastructure grows to support a variety of services, its legacy protocols are being overloaded with new functions such as traffic engineering. Today, operators engineer such capabilities through clever, but manual parameter tuning. In this paper, we propose a back-end support tool for large-scale parameter configuration that is based on efficient parameter state space search techniques and on-line simulation. The framework is useful when the network protocol performance is sensitive to its parameter settings, and its performance can be reasonably modeled in simulation. In particular, our system imports the network topology, relevant protocol models and latest monitored traffic patterns into a simulation that runs on-line in a network operations center (NOC). Each simulation evaluates the network performance for a particular setting of protocol parameters. We propose an efficient large-dimensional parameter state space search technique called “recursive random search (RRS).” Each sample point chosen by RRS results in a single simulation. An important feature of this framework is its flexibility: it allows arbitrary choices in terms of the simulation engines used (e.g., ns-2, SSFnet), network protocols to be simulated (e.g., OSPF, BGP), and in the specification of the optimization objectives. We demonstrate the flexibility and relevance of this framework in three scenarios: joint tuning of the RED buffer management parameters at multiple bottlenecks, traffic engineering using OSPF link weight tuning, and outbound load-balancing of traffic at peering/transit points using BGP LOCAL_PREF parameter.      

Genetic algorithm‐based routing method for enhanced video delivery over software defined networks

Video streaming has emerged as a killer application in today's Internet, delivering a tremendous amount of media contents to millions of users at any given time. Such a heavy traffic load demands an effective routing method. In this paper, an effective routing method, named GA‐SDN, is developed based on software defined network (SDN) technique. To facilitate the researchers in this field to evaluate the video delivery quality over SDN, an evaluation framework and its associated source codes are provided. The framework integrates the H.264 Scalable Video coding streaming Evaluation Framework (SVEF) with the Mininet emulator. Through this framework, video processing researchers can evaluate their proposed coding algorithms in an SDN‐enabled network emulator, while network operators or executives can evaluate the impact of real video streams on the developing network architectures or protocols. Experiment results demonstrate the usefulness of myEvalSVC_SDN and prove that GA‐SDN outperforms traditional Bellman‐Ford routing algorithm in terms of packet drop rate, throughput, and average peak signal‐to‐noise ratio.     

Enabling novel premium service classes in DiffServ over MPLS‐enabled network

Network resources dimensioning and traffic engineering influence the quality in provisioned services required by the Expedited Forwarding (EF) traffic in production networks established through DiffServ over MPLS‐enabled network. By modeling EF traffic flows and the excess of network resources reserved for it, we derive the range of delay values which are required to support these flows at DiffServ nodes. This enables us to develop an end‐to‐end (e2e) delay budget‐partitioning mechanism and traffic‐engineering techniques within a framework for supporting new premium QoS levels, which are differentiated based on e2e delay, jitter and loss. This framework enables ingress routers to control EF traffic flow admission and select appropriate routing paths, with the goal of EF traffic balancing, avoiding traffic congestion and getting the most use out of the available network resources through traffic engineering. As a result, this framework should enable Internet service providers to provide three performance levels of EF service class to their customers provided that their network is DiffServ MPLS TE aware. Copyright © 2008 John Wiley & Sons, Ltd.     

Transmission of adaptive MPEG video over time-varying wireless channels: modeling and performance evaluation

Wireless channels are characterized by high time-varying bit-error rates (BERs). To cope with this problem, several adaptive forward-error-correction (AFEC) schemes have been proposed in the literature. They work locally at the wireless link, adding a variable amount of redundancy to the transmitted data in order to maintain the packet error rate below an acceptable level. However, when such schemes are utilized, the bandwidth offered to the applications changes when channel conditions change. In this paper, the effects of these bandwidth variations are investigated in the case of real-time Motion Picture Experts Group (MPEG) video transmission. The MPEG encoder is controlled in order to adapt its emission rate to the current bandwidth offered by the wireless link. To this end, the encoding quality is diminished by the source rate controller when the transmission rate has to be decreased due to an increase in the channel BER, whereas it is improved when the transmission rate can be increased due to a decrease in the channel BER. A Markov-based model, denoted as SBBP/SBBP/1/K, has been introduced to model the scenario being considered. The analytical framework allows evaluation of the performance of the system and can be used to optimize the design of a video transmission system for wireless channels, providing the instruments to derive the tradeoff between information corruption in the wireless channel and MPEG video encoding quality.     

Network parameters awareness for routing discovery in autonomic optical Internet

The article proposes a network parameter-awareness (NPA) method and applies it to routing discovery algorithms in autonomic optical Internet. This NPA method can perceive the main parameters of the network, such as delay, jitter and traffic, which can represent the current situation of the network. And these parameters enable network to determine the appropriate nodes for routing discovery. The simulation results of evaluating performance of a network with NPA method and its routing applications show that the method and its applications in routing improve the performance of the network significantly with quality of service (QoS) guaranteed.     

Smoothing and transporting video in QoS IP networks

Real‐time traffic such as voice and video, when transported over the Internet, demand stringent quality of service (QoS) requirements. The current Internet as of today is still used as a best effort environment with no quality guarantees. An IP‐based Internet that supports different QoS requirements for different applications has been evolving for the past few years. Video streams are bursty in nature due to the instant variability of the video content being encoded. To help mitigate the transport of bursty video streams with minimal loss of information, rate‐adaptive shapers (RASs) are usually being used to reduce the burstiness and therefore help preserve the desired quality. When transporting video over a QoS IP network, each stream is classified under a specific traffic profile to which it must conform, to avoid packet loss and picture quality degradation. In this paper we study, evaluate and propose RASs for the transport of video over a QoS IP network. We utilize the encoding video parameters for choosing the appropriate configuration needed to support the real‐time transport of Variable Bit Rate (VBR) encoded video streams. The performance evaluation of the different RASs is based on the transport of MPEG‐4 video streams encoded as VBR. The performance is studied based on looking at the effect of various parameters associated with the RASs on the effectiveness of smoothing out the burstiness of video and minimizing the probability of packet loss. Copyright © 2005 John Wiley & Sons, Ltd.     

An analytical framework for the design of intelligent algorithms for adaptive-rate MPEG video encoding in next-generation time-varying wireless networks

Adaptive rate video encoding is required to maximize efficiency when wireless links are involved in the communication. In fact, wireless channels are characterized by high, time-varying bit error rates. To cope efficiently with this problem adaptive forward error correction schemes have been proposed. These schemes introduce an amount of redundancy dependent on the channel conditions. Accordingly, the bandwidth available at the application layer changes: it increases when channel conditions improve, and decreases when channel conditions worsen. Obviously, the encoding parameters must be tuned to adapt the video source transmission rate to the available bandwidth. This adaptation is achieved by means of appropriate feedback laws, which are relationships between the encoding parameters to be used and other variables representing the state of the system. An analytical framework is introduced which can be used for the design of the feedback laws. To this purpose both the channel and the video source are modeled by means of Markov models. The resulting model of the whole system is denoted as SBBP/SBBP/1/K. Analysis is derived which allows to evaluate the most significant performance measures and, therefore, to design optimal feedback laws.