无线Mesh网中时延约束抖动优化的多路径流量分配算法

针对无线mesh网中多媒体应用的特点,研究多路径传输中路径时延满足约束且路径间抖动最小化的路径流量分配问题。首先,基于网络演算理论分析了数据分组在无线mesh网单路径传输中产生的队列时延,推导出单路径传输的时延上界以及多路径传输中的路径间时延抖动上界,并得到满足时延约束的路径最大容许流入速率;然后,基于时延及其抖动上界,提出满足时延约束抖动优化的路径流量分配算法DCJOTA,该算法根据路径最大容许流入速率按比例分配各路径流量,同时尽可能减小路径间的时延抖动;最后,分析了算法DCJOTA的可行性及其实现方法,并在NS2网络模拟器中验证了该算法的有效性。仿真实验表明,与AOMDV协议相比,集成了DCJOTA算法的多路径路由协议DCJO-AOMDV协议在时延及其抖动方面具有更好的表现:端到端平均时延降低3.9%,端到端平均时延抖动减小24.5%。另外,DCJOTA算法带来协议复杂性略微增加,DCJO-AOMDV协议下的网络吞吐量下降1.7%。     

IP网QoS保证及3G R4版本IP承载网规划

众所周知,QoS保证一直是IP网没有得到很好解决的问题,而3G R4版本如果采用IP承载,必然面临这一关键问题。为了更好地探讨3G R4版本IP承载网实现QoS保证的技术,本文将首先分析一下IP承载网端到端的QoS。一、IP承载网端到端QoS分析影响端到端QoS的因素有很多,主要包括呼叫建立时间、丢包率、抖动(时延变化)和端到端时延等,其中影响到IP承载网的质量指标体现在丢包率、抖动和时延上。为了保证长途话音的服务质量,要求IP承载网络的抖动应小于10m s,丢包率小于1%。在IP电话系统中,端到端的时延要求分成四个等级:小于等于150m s(最好)…     

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。     

基于OMNeT++的大规模确定性网络仿真实践

时间敏感网络（TSN）和确定性网络（DetNet）是确定性网络整体解决方案的重要组成部分,目前产业界还缺少这两种技术在大规模联合组网方面的研究。在OMNeT++仿真平台基于时隙调度实现了TSN和DetNet联合组网的确定性转发,并打通了控制器和仿真系统的通道,实现了转控协同的确定性仿真网络。重点介绍了对于仿真系统的改造,以及基于此的大规模跨域场景下确定性业务的端到端时延和抖动性能研究。结果表明,在大规模跨域场景下,时隙调度的机制可以提供端到端的确定性业务保障。     

面向大规模确定性网络的全局循环排队与转发机制

时延抖动敏感的工业控制和远程驾驶应用驱动网络由尽力而为服务向确定性转发服务转变。为满足确定性应用时延抖动的有界需求,工业界和学术界在逐流过滤整形和队列转发控制方面进行了大量研究,实现了小规模以太网环境下轻载稳定流的微秒级时延和亚微秒级抖动保障。多数研究因较少考虑长距离网络拓扑变化、传输时延波动和短时突发流量过载等影响,难以满足大规模骨干网场景下的确定性要求。在循环排队转发（cyclic queuing and forwarding,CQF）基础上,考虑了源目的路径特性和确定性流量强度因素,提出了全局循环排队转发三队列（global cyclic queuing and forwarding 3-queue,GCQF-3）,并在基于OMNeT++搭建的确定性网络仿真系统中,比较了CQF、CQF-3和GCQF-3的转发控制机制,实验表明GCQF-3能根据确定性流量状态及时调整门控时机和排队优先级,在时延、抖动和网络利用率等方面都达到最佳。     

移动IP网络承载实时业务的4个关键

移动IP网络毕竟是一项新的技术及应用,特别在对实时业务的承载上,有其独有的特色。不同的业务有着不同的业务要求,具体体现在速率、时延、时延抖动以及丢包率等方面。从IP网络的角度,一般采用QoS来区分不同的业务需求。业界第一个实现移动网络下采用IP机制承载实时业务的cdma2000 1XEV-DO RevA网络,是一个端到端有QoS保证的网络,对用户的QoS有严格的定义和分类,有用户间的QoS定义、用户内不同业务QoS的定义以及对应不同的数据流的定义。     

IP QoS服务模型及其相关问题的探讨

1 采用QoS机制的必要性 传统的IP网络只提供“尽力而为”的数据传输能力。随着网络上主机数量的不断增加,网络服务的需求将超过网络提供的能力,从而造成传输时延变化（抖动）、传输时延过大甚至引起分组丢失,也就是说出现了网络拥塞。网络拥塞对一些Internet应用（如电子邮件,文件传输和Web应用）一般不会造成太大影响,但对传输时延要求比较苛刻的实时应用（如多媒体业务）及大多数双向通信业务（如电话业务）却是不能容忍的。     

基于统计网络演算的端到端时延上界研究

为分析融合网络端到端统计时延性能,根据最小加代数理论中的统计网络演算相关知识,提出了一种改进的统计服务曲线并应用GPS调度系统模型,然后利用网络演算理论求解串型网络节点统计时延上界。数值分析结果表明,改进的调度模型有效改善了自相关特性对时延上界所造成的影响,比未改进的模型相比统计时延上界更具有良好的紧致性,为在融合网络环境中对端到端节点延迟上界研究提供有效控制、调度和管理具有一定的参考价值。     

网络QoS特性分析

服务质量（ＱｏＳ）是指在网络连接中由业务所决定的性能度量，如带宽、时延或连接中传送数据所允许的消息丢失率。ＱｏＳ需要通过新的连接发起时的接入控制机制获得保证。 服务质量的量化指标 用户感知到的服务质量，即传输的时延。时延上要从以下方面对不同服务产生不同的影响：端到端时延（Ｅｎｄ－ｔｏ－ｅｎｄ　ｄｅｌａｙ）；时延变化（即抖动，ｊｉｔｔｅｒ）。交互式的实时业务（如语音通信等）对端到端时延和抖动很敏感。非交方式的实时业务（如单向广播等）对端到端时延不敏感，但对抖动敏感。非实时业务往往对时延不敏感，但由于…     

基于路径优先度的VoIP中继选择算法

在覆盖网络中为VoIP业务计算端到端路径时,需要保证该路径的时延小于150 ms;由于覆盖网络中反三角现象普遍存在,因此,为了保证VoIP业务时延要求,应用中继技术计算从源节点通过中继节点到目的节点的路径,取代端到端路径,成为覆盖网络提供VoIP业务的关键技术。基于已有中继技术存在的部署难、未考虑路径差异及结果不是最优等问题,提出基于路径优先度的VoIP中继选择算法,该算法首先提出路径优先度的概念,描述时延和默认IP路径的差异性,再基于路径优先度构建中继路由表,在中继路由表中挑选最优中继节点;提出基于路径优先度的两跳中继选择算法进一步减少转发时延。仿真证明所提算法能够减少VoIP业务的传输时延,提升VoIP业务的用户体验。     

端到端确定性网络架构和关键技术

新型业务如远程控制、虚拟/增强现实等对网络提出了更高的带宽和时延要求,传统网络已经难以满足。面向6G的网络演进,需要端到端确定性网络技术保障。针对当前技术标准研究缺乏对于端到端时延保障的全局考虑,指出了确定性网络需要同时解决单跳和全局的时延问题,提出了端到端确定性网络架构,以及跨网络域和跨协议层的确定性网络技术方案。最后,针对当前确定性网络应用部署的难点,提出了相应的推进建议。     

Packet sequencing: a deterministic protocol for QoS in IP networks

We describe a deterministic protocol for routing delay and loss-sensitive traffic through an IP network. Unlike traditional approaches, the method described here - packet sequencing - does not rely on queue management. Instead, it uses a temporally-based deterministic protocol to coordinate and switch IP packets on a systemwide basis. As a result, end-to-end throughput is guaranteed, without packet loss, loss variance, or accumulated performance impairment; additionally, end-to-end delay is minimized, and jitter is essentially eliminated. We also show that packet sequencing can complement conventional IP networks: sequencing does not negate the use of queue management QoS methods that are the subject of considerable ongoing study. This article describes the fundamental approach, issues associated with scalability, illustrative performance in the context of storage networking, and attributes related to the security and reliability of IP networks.     

Internet Multicasting of IPTV With Essentially-Zero Delay Jitter

A technology for multicasting packetized multimedia streams such as IPTV over the Internet backbone is proposed and explored through extensive simulations. An RSVP or DiffServ algorithm is used to reserve resources (i.e., bandwidth and buffer space) in each packet-switched IP router in an IP multicast tree. Each IP router uses an Input-Queued (IQ) switch architecture with unity speedup. A recently proposed low-jitter scheduling algorithm is used to pre-compute a deterministic transmission schedule for each IP router. The IPTV traffic will be delivered through the multicast tree in a deterministic manner, with bounds on the maximum delay and jitter of each packet (or cell). A playback buffer is used at each destination to filter out residual network jitter and deliver a very low-jitter video stream to each end-user. Detailed simulations of an IPTV distribution network, multicasting 75 high-definition video streams over a fully-saturated IP backbone are presented. The simulations represent the transmission of 129 billion cells of real video data and where performed on a 160-node cluster computing system. In the steady-state, each IP router buffers approx. 2 cells (128 bytes) of video data per multicast output-port. The observed delay jitter is zero when a playback buffer of 15 milliseconds is used. All simulation parameters are presented.      

Technical Analysis on Application Layer Error Control in IPTV

When they provide Internet Protocol Television (IPTV) services,IP networks face difficulties such as reduced transmission bandwidth,delay,jitter,and packet loss.A terminal-centric solution can improve the image quality by adding control policies like congestion control and error control to the receiving terminal and source code.An end-to-end application layer reliability method is usually required to satisfy IPTV requirements in the IP network.Application Layer Forward Error Correction (AL-FEC) technology s...     

面向业务和网络协同的未来IP网络架构演进

IP技术作为互联网的技术基础,在今后5～10年如何发展是通信行业面临的关键问题。分析了传统IP网络架构的成功与不足,提出未来IP网络既要继承传统IP技术的设计原则,比如端到端原则和分层解耦原则,又要解决现有IP网络中业务和网络过于割裂的问题。从网络演进和未来业务需求的角度出发,认为未来IP网络的核心问题是要加强业务和网络的协同,提出了改进后的IP网络设计原则“横向：服务化网络赋能的端到端架构,纵向：智能控制面支撑的分层模型”,并且基于此原则提出了未来网络的参考架构。     

应用RSVP提高视频会议系统业务传输质量

传统的IP网络基于尽力而为的设计思想,不对上层承载业务提供任何服务质量保障。而视频会议应用对服务质量要求相对严格,延迟及延迟抖动会降低用户的应用体验。针对基于IP网络的视频会议系统中缺少服务质量保障的问题,提出了在视频会议系统中部署资源预留协议以提供服务质量保障的方案,并且在Vxworks系统中完成资源预留协议的开发。为了验证方案效果,分别在中度负载和重度负载的情况下对业务性能进行测试。测试结果表明,该方案能够为视频会议系统提供有效的服务质量保障。     

SMS-MIPv6: An End-to-End Mechanism for IPv6 Mobility Management in Mobile Networks

As IP has been extended from core networks to access networks, a mobile network can be considered as an overlay of a traditional cellular network and an IP network. SMS-MIPv6 attempts to integrate mobility management of these two kinds of networks. The basic idea behind SMS-MIPv6 is to exploit existing mobility management in the cellular network (i.e. in the form of well-defined short messages) to locate a Mobile Terminal (MT) in the IPv6 network. We should emphasize that the motivation of SMS-MIPv6 is not to replace or optimize existing mature mobility management schemes. On the contrary, as an entirely end-to-end mechanism for IPv6 mobility management, it provides an alternative mechanism for free peer-to-peer applications such as Voice over IP (VoIP) without support from mobile network operators. We describe the implementation of SMS-MIPv6 in detail and analyze its performance. The evaluation results show that SMS-MIPv6 achieves acceptable performance so that it can be deployed in most current mobile networks. It performs best in terms of signaling cost, data traffic overhead compared with Mobile IPv6 (MIPv6) and Proxy MIPv6 (PMIPv6). Moreover, SMS-MIPv6 can reduce the handover latency significantly, although it is considered as a mobility management scheme for global mobility. However, it increases the session initialization latency due to hybrid binding through the cellular network.     

Survey of the deterministic network

The current Internet faces a huge increase in video traffic and industrial machine applications,causing a lot of congestion and packet delays.Besides,many network applications,such as industrial Internet,telemedicine,driverless,VR games,etc,require a latency from 1 to 10 milliseconds and jitters within microsecond.But traditional network can only reduce the end-to-end latency to tens of milliseconds.Under this background,how to transit from “best effort” to“punctuality,accuracy”,controlling the end-to-end delay has become a hot field of global network research.The application scenarios and requirements of the deterministic network were introduced,and the main research results of the current parties were described as well as the trend of development and core issues were summarized.It hopes to provide a reference and help for the research in this field.     

An analytical model for jitter in IP networks

Traditionally, IP network planning and design is mostly based on the average delay or loss constraints which can often be easily calculated. Jitter, on the other hand, is much more difficult to evaluate, but it is particularly important to manage the QoS of real-time and interactive services such as VoIP and streaming video. In this paper, we present simple formulas for the jitter of Poisson traffic in a single queue that can be quickly calculated . It takes into account the packets delay correlation and also the correlation of tandem queues that have a significant impact on the end-to-end jitter. We then extend them to the end-to-end jitter of a tagged stream based on a tandem queueing network. The results given by the model are then compared with event-driven simulations. We find that they are very accurate for Poisson traffic over a wide range of traffic loads and more importantly that they yield conservative values for the jitter so that they can be used in network design procedures. We also find some very counter-intuitive results. We show that jitter actually decreases with increasing load and the total jitter on a path depends on the position of congested links on that path. We finally point out some consequences of these results for network design procedures.