Expedited forwarding end‐to‐end delay and jitter in DiffServ

The scheduling disciplines and active buffer management represent the main components employed in the differentiated services (DiffServ) data plane, which provide qualitative per‐hop behaviors corresponding to the QoS required by supported traffic classes. In the first part of this paper, we compute the per‐hop delay bound that should be guaranteed by the different multiservice scheduling disciplines, so that the end‐to‐end (e2e) delay required by expedited forwarding (EF) traffic can be guaranteed. Consequently, we derive the e2e delay bound of EF traffic served by priority queuing–weighted fair queuing (PQ–WFQ) at every hop along its routing path. Although real‐time flows are principally offered EF service class, some simulations on DiffServ‐enabled network show that these flows suffer from delay jitter and they are negatively impacted by lower priority traffic. In the second part of this paper, we clarify the passive impact of delay jitter on EF traffic, where EF flows are represented by renewal periodic ON–OFF flows, and the background (BG) flows are characterized by the Poisson process. We analyze through different scenarios the jitter effects of these BG flows on EF flow patterns when they are served by a single class scheduling discipline, such as first‐input first‐output, and a multiclass or multiservice scheduling discipline, such as static priority service discipline. As a result, we have found out that the EF per‐hop behaviors (PHBs) configuration according to RFCs 2598 and 3246 (IETF RFC 2598, June 1999; RFC 3246, IETF, March 2002) cannot stand alone in guaranteeing the delay jitter required by EF flows. Therefore, playout buffers must be added to DiffServ‐enabled networks for handling delay jitter problem that suffers from EF flows. Copyright © 2008 John Wiley & Sons, Ltd.     

基于剩余路径跳数的动态优先调度实现区分服务EF PHB

Internet区分服务（DiiffServ）中EF PHB(Expedited Forwarding Per Hop Behavior)提供严格的端到端延迟保证，其实现机制和性能是当前研究的热点。随着可扩展性成为核心网络考虑的关键因素，一般用简单的FIFO高度实现EF PHB。FIFO实现问题在于最坏的端到端延迟与流经历的最大跳数成正比，结果降低了网络最坏延迟性能，并影响了整个网络的总体利用率。文章在分析并比较FIFO实现以及考虑流跳数因素的绝对跳数优先（HBAP）实现、相对跳数优先（HBRP）实现的延迟性能基础上，提出了用基于剩余路径跳数的动态优先（DHBP）调度实现EF PHB。理论分析和实验结果表明，基于剩余路径跳数的动态优先调度算法可以平衡不同跳数流的端到端延迟性能，从而减小网络最坏的端到端延迟，并有效地提高了网络 选用率，最坏延迟性能明显优于FIFO和绝对跳数优先调度，与性能最优的相对跳数优先调度相似，并将计算复杂度降为0(1)。     

The effect of traffic shaping in efficiently providing end-to-end performance guarantees

This paper reports new results concerning the capabilities of a family of service disciplines aimed at providing per-connection end-to-end delay (and throughput) guarantees in high-speed networks. This family consists of the class of rate-controlled service disciplines, in which traffic from a connection is reshaped to conform to specific traffic characteristics, at every hop on its path. When used together with a scheduling policy at each node, this reshaping enables the network to provide end-to-end delay guarantees to individual connections. The main advantages of this family of service disciplines are their implementation simplicity and flexibility. On the other hand, because the delay guarantees provided are based on summing worst case delays at each node, it has also been argued that the resulting bounds are very conservative which may more than offset the benefits. In particular, other service disciplines such as those based on Fair Queueing or Generalized Processor Sharing (GPS), have been shown to provide much tighter delay bounds. As a result, these disciplines, although more complex from an implementation point-of-view, have been considered for the purpose of providing end-to-end guarantees in high-speed networks. In this paper, we show that through proper selection of the reshaping to which we subject the traffic of a connection, the penalty incurred by computing end-to-end delay bounds based on worst cases at each node can be alleviated. Specifically, we show how rate-controlled service disciplines can be designed to outperform the Rate Proportional Processor Sharing (RPPS) service discipline. Based on these findings, we believe that rate-controlled service disciplines provide a very powerful and practical solution to the problem of providing end-to-end guarantees in high-speed networks.     

Per-domain packet scale rate guarantee for expedited forwarding

The expedited forwarding per-hop behavior (EF PHB) was recently replaced by a new definition, called packet scale rate guarantee (PSRG), under the Differentiated Services (DiffServ) framework. This replacement raises two challenges. One is the implementation of a PSRG server. Another is the provision of per-domain PSRG. Specifically, for the former, an open issue is whether hierarchical schedulers can provide PSRG; for the latter, it is not clear whether and how per-domain PSRG can be provided in the presence of flow aggregation. Since, in DiffServ networks, flow aggregation is a natural phenomenon and hierarchical scheduling is high-likely desired, these two challenges become even more critical. To address the first challenge, we introduce a new concept called latency-rate worst-case service guarantee (LR-WSG). We prove that, if a server provides LR-WSG, it also provides PSRG. We show that many well-known schedulers support LR-WSG, which include not only one-level schedulers but also their hierarchical versions. To address the second challenge, we first prove that PSRG can be extended from per-node to per-domain if no flow aggregation is performed. The proof is notable in that it depends solely on the concept of PSRG itself. We then investigate the provision of per-domain PSRG in presence of flow aggregation. We propose to use packet scale fair aggregator (PSFA) to aggregate flows. We show that, with PSFA, per-domain PSRG can be provided in spite of flow aggregation. We finally provide a brief discuss on the viability of using PSFA in DiffServ networks and define an expedited forwarding per-domain behavior (EF PDB).     

光突发交换网络中Jitter-EDF调度算法的研究

提出了一种应用于OBS网络中的Jitter-EDF调度算法来减小时延抖动对实时性突发QoS的影响,设计了基于该算法的OBS网络中核心路由器的主要结构,讨论了此算法实现的可能性,并推导出端到端时延的上下限和时延抖动的上限,最后结合具体网络拓扑进行了仿真和分析。结果表明,Jitter-EDF调度算法能够将时延抖动的幅度平均降低到30%～40％,有效保证了实时性应用对时延抖动的要求。     

Group priority scheduling

We present an end-to-end delay guarantee theorem for a class of guaranteed deadline (GD) servers. The theorem can be instantiated to obtain end-to-end delay bounds for a variety of source control mechanisms and GD servers. We then propose the idea of group priority, and specialize the theorem to a subclass of GD servers that use group priority in packet scheduling. With the use of group priority, the work of packet schedulers can be substantially reduced. We work out a detailed example, for the class of burst scheduling networks, to illustrate how group sizes can be designed such that the worst case end-to-end delay of application data units in a real-time flow is unaffected by the use of group priority. Group priority also can be used in packet schedulers that provide integrated services (best effort as well as real-time services) to achieve statistical performance gains, which we illustrate with empirical results from simulation experiments     

大规模确定性网络转发技术

提出了一种适用于大规模网络部署的3层转发技术——LDN（large-scale deterministic network）,在保留传统IP转发技术统计复用的优势基础之上,LDN技术可实现对端到端时延上界及抖动上界的严格保证,为5G uRLLC（ultra-reliable low-latency communication）切片、工业互联网等未来应用场景提供网络服务支持。通过仿真实验对比了在相同网络环境下,传统IP及确定性IP在端到端最差时延及抖动上的差异,证明了LDN技术的有效性。     

保障流媒体时间结构的网络时延分析

流媒体同步对端到端时延和时延抖动提出了确定的要求,而终端抖动缓存一方面能消除时延抖动的影响,一方面却增加了端到端时延,流媒体同步保障对网络时延的要求不明确。论文从概率保障流媒体同步的角度,确定了保障流媒体同步的抖动缓存容量范围,提出了流媒体同步网络保障的充分条件,针对基于Internet VoIP(Voice over IP)业务的实际网络测试结果,给出了应用流媒体同步网络保障充分条件进行同步保障评价的应用实例并验证了其正确性。     

Service disciplines for guaranteed performance service inpacket-switching networks

While today's computer networks support only best-effort service, future packet-switching integrated-services networks will have to support real-time communication services that allow clients to transport information with performance guarantees expressed in terms of delay, delay jitter, throughput, and loss rate. An important issue in providing guaranteed performance service is the choice of the packet service discipline at switching nodes. In this paper, we survey several service disciplines that are proposed in the literature to provide per-connection end-to-end performance guarantees in packet-switching networks. We describe their mechanisms, their similarities and differences and the performance guarantees they can provide. Various issues and tradeoffs in designing service disciplines for guaranteed performance service are discussed, and a general framework for studying and comparing these disciplines are presented     

航空电子系统中FC网络的仿真和延迟估计

对采用单交换机光纤通道(FC)网络的航空电子系统进行了描述,建立了网络的排队模型.根据M/G/1模型对FC网络的队列长度、服务时间等参数进行了分析和估计.在OPNET平台下对4种周边节点数不同的FC网络进行了仿真;通过仿真分析了FC网络端到端延迟和网络节点数关系,并对评估的数学模型进行了修正;研究了网络中的延迟"抖动"...     

Real-time multimedia scheduling policies for end-to-end delayjitter and loss guarantees across ATM satellite systems

We propose a simple first-in first-out (FIFO)-based service protocol which is appropriate for a multimedia ATM satellite system. The main area of interest is to provide real-time traffic with upper bounds on the end-to-end delay, jitter, and loss experienced at various service queues within a satellite network. Various service protocols, each based on a common underlying strategy, are developed in light of the requirements and limitations imposed at each of the satellite's subsystems. These subsystems include the uplink (UL) earth station (ES) service queue, on-board processing (OBP) queues, and the downlink (DL) ES service queue feeding into a wireline ATM network or directly to an end-user application. Numerous network simulation results demonstrate the tractability, efficiency, and versatility of the underlying service discipline. Key features of our strategy are its algorithmic and architectural simplicity, its non-ad-hoc scheduling approach, and its unified treatment of all real-time streams at all service queues. In addition, the delay and jitter bounds are uncoupled. In this way, end-to-end jitter can be tightly controlled even if medium access requires long indeterminate waiting durations     

A Dual Feedback Approach of Request-Based Bandwidth Allocation for Real-Time Service in Broadband Wireless Access Networks

We propose an efficient request-based uplink bandwidth allocation algorithm for variable-rate real-time service in broadband wireless access networks. By introducing a notion of target delay under the framework of dual feedback, the proposed algorithm can regulate delay while minimizing delay jitter and bandwidth waste.     

UMTS核心网中基于区分服务的QoS控制模型

3G新业务的发展，要求UMTS提供端到端QoS控制。文章构建了在UMTS核心网中为不同业务类提供QoS保证的区分服务模型，提出了从UMTS业务类到DiffServ域服务等级的映射方案，设计了一种新的队列调度算法，采用优先级和分离机制，在流量调整器配合下可满足不同业务类的QoS要求。最后，通过模拟实验证明了模型的有效性。     

Policies for allocating performance impairment budgets among multiple IP providers

An important technical aspect of achieving end-to-end quality of service (QoS) in next generation networks refers to allocation of the total performance impairment budget (delay, jitter, packet loss) among multiple providers. In this article, we first propose a generic policy for allocating per-domain impairment budgets, relying on the set of performance metrics from service request and the rules for their composition on the multi-domain path. The objective is to provide end-to-end QoS through the set of heterogeneous domains, with different QoS models and definitions of service classes. The allocation of impairment budgets among multiple domains is then closely related to mapping of service classes between providers and selection of the most appropriate class for particular service in each domain. Based on the generic policy, we further derive examples of specific policies and evaluate them with respect to fulfillment of QoS objectives, fairness, adaptability and scalability. Evaluation tool implements a policy-based conformance matching scheme, which enforces selection of the domain class that most tightly matches with the required QoS.     

IP QoS across multiple management domains: practical experiences from pan-European experiments

GEANT is the pan-European 10 Gbs network interconnecting European national research and educational networks (NRENs). A Premium IP service based on the DiffServ EF PHB has been specified and implemented for this environment to provide quality of service to selected user groups on a Europe-wide scale. Basic features of Premium IP are described, and results from early experiments in the production networks of GEANT and the NRENs are presented. Next steps are proposed for achieving a fast and wide availability of Premium IP in the European research networks.     

Congestion control for multimedia services

The problem of congestion control in high-speed networks for multimedia traffic, such as voice and video, is considered. It is shown that the performance requirements of high-speed networks involve delay, delay-jitter, and packet loss. A framing congestion control strategy based on a packet admission policy at the edges of the network and on a service discipline called stop-and-go queuing at the switching nodes is described. This strategy provides bounded end-to-end delay and a small and controllable delay-jitter. The strategy is applicable to packet switching networks in general, including fixed cell length asynchronous transfer mode (ATM), as well as networks with variable-size packets     

Calculation of packet jitter for non-poisson traffic

The packet delay variation, commonly called delay jitter, is an important quality of service parameter in IP networks especially for real-time applications. In this paper, we propose the exact and approximate models to compute the jitter for some non-Poisson FCFS queues with a single flow that are important for recent IP network. We show that the approximate models are sufficiently accurate for design purposes. We also show that these models can be computed sufficiently fast to be usable within some iterative procedure, e.g., for dimensioning a playback buffer or for flow assignment in a network.     

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.     

Seamless SIP-based mobility for multimedia applications

Application-level protocol abstraction is required to support seamless mobility in next-generation heterogeneous wireless networks. Session initiation protocol (SIP) provides the required abstraction for mobility support for multimedia applications in such networks. However, the handoff procedure with SIP suffers from undesirable delay and hence packet loss in some cases, which is detrimental to applications like voice over IP (VoIP) or streaming video that demand stringent quality of service (QoS) requirements. In this article we present a SIP-based architecture that supports soft handoff for IP-centric wireless networks. Soft handoff ensures that there is no packet loss and that the end-to-end delay jitter is kept under control.     

A Probabilistic Study of the Delay Caused by Jittering in Wireless Flooding

Systematic packet collisions constitute a major problem in wireless flooding, which is a key mechanism for information dissemination in wireless mesh and multi-hop ad hoc networks. Since this cannot be solved only through classic MAC collision avoidance mechanisms, the IETF has proposed and standardized in RFC 5148 jittering techniques to handle it. These techniques are widely used in protocols for wireless communication such as OLSR, AODV or LOAD, and have proven useful for reducing collisions. They lead however some undesirableside effects that may harm substantially the flooding performance. To the best of our knowledge, no research effort has been deployed to understand and analyze these effects. This paper addresses this issue. It motivates and introduces a theoretical model of flooding with jitter in a wireless interface, as specified in RFC 5148, and explores the probabilistic characterization of additional flooding delay caused by jitter. It mostly provides two analytical bounds for the per-interface additional jitter delay. Presented results, which are validated by way of a discrete-event simulation, enable a better understanding of the performance trade-offs (between packet collisions and additional delay, in particular) underlying the use of jitter in wireless flooding.