Internet protocols for multimedia communications. II. Resourcereservation, transport, and application protocols

Part I (ibid., July-Oct. 1997) surveyed the evolution of Internet protocols and applications and described the Internet protocol IPv6 in detail. This part discusses new developments at the upper layers that support real-time Internet multimedia, such as audio and video conferencing and shared whiteboard applications. Application-level framing (ALF), proposed in 1990 for protocol and application design, now forms the basis for many new Internet protocols and applications, including Real-time Transport Protocol (RTP) and Mbone multimedia applications. RTP supports real-time applications that adapt to changing network situations to maintain the quality of service (QoS). The Resource Reservation Protocol (RSVP) provides new Internet services with higher quality than best-effort by means of resource reservations     

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     

End-to-end quality-of-service coordination for mobile multimedia applications

Transmitting real-time multimedia streams over heterogeneous mobile networks is a challenging task. Variation in network and system conditions can dramatically affect application performance. When providing end-to-end quality-of-service (QoS) multiple system facets should be coordinated: orchestration of local and peer resources, reservation of network resources, adaptation of multimedia streams, etc. This paper presents an end-to-end negotiation protocol (E2ENP) for negotiating and coordinating QoS on an end-to-end basis both at application and network layer. Based on a flexible extensible markup language (XML) model and extending SDPng concepts, the protocol enables the negotiation of system capabilities and allows provider-services to effectively influence the negotiation process. The aim of the E2ENP design is to optimize the efficiency of multimedia call setup and reduce the time for QoS renegotiations, whenever vertical handovers or spontaneous network reconfigurations occur. The basic protocol is presented, together with implementation and measurement results, stemming from several studies on current and future third-generation/fourth-generation scenarios.     

QoS provisioning for multimedia wireless networks

As we move towards IP-based multimedia wireless networks with voice, video and data convergence, quality of service (QoS) provisioning will become an increasingly challenging task. One implication is that greater emphasis on managing the call admission and overall network resources will be needed. This paper presents a conservative and adaptive quality of service (CAQoS) framework for provisioning the QoS for both real-time and non-real-time traffic in a multimedia wireless network. Unlike most conventional schemes, which gradually scale down the bandwidth of ongoing connections to accommodate new connection/hand-off requests, CAQoS introduces an early scaling-down of bandwidth for new connections based on a designated provisioning model. The performance of a CAQoS system is evaluated through simulations of a realistic wireless environment. Simulation results show that CAQoS meets our design goals and outperforms conventional schemes.     

A Bandwidth Control Method Providing Entrance QoS for Multimedia Communication

1 Motivation Nowadays , the QoS provided by the wideband IPnetwork starts at an edge-router and ends at anotheredge-router .Therefore ,the trunk QoSisfinished,suchas RSVP[1]and DiffServ[2].In order to complete theend-to-end QoS,the mechanism of entrance QoS (thebandwidth controller) is introduced. As Fig.1 shows ,if the bandwidth controller circledby the dashed frame is not used,the Ethernet is con-nected with the router directly through 100 Mbit/sbandwidth. Then this router is connect…     

Enabling seamless multimedia wireless access through QoS‐based bandwidth adaptation

Effective support of real‐time multimedia applications in wireless access networks, viz. cellular networks and wireless LANs, requires a dynamic bandwidth adaptation framework where the bandwidth of an ongoing call is continuously monitored and adjusted. Since bandwidth is a scarce resource in wireless networking, it needs to be carefully allocated amidst competing connections with different Quality of Service (QoS) requirements. In this paper, we propose a new framework called QoS‐adaptive multimedia wireless access (QoS‐AMWA) for supporting heterogeneous traffic with different QoS requirements in wireless cellular networks. The QoS‐AMWA framework combines the following components: (i) a threshold‐based bandwidth allocation policy that gives priority to handoff calls over new calls and prioritizes between different classes of handoff calls by assigning a threshold to each class, (ii) an efficient threshold‐type connection admission control algorithm, and (iii) a bandwidth adaptation algorithm that dynamically adjusts the bandwidth of an ongoing multimedia call to minimize the number of calls receiving lower bandwidth than the requested. The framework can be modeled as a multi‐dimensional Markov chain, and therefore, a product‐form solution is provided. The QoS metrics—new call blocking probability (NCBP), handoff call dropping probability (HCDB), and degradation probability (DP)—are derived. The analytical results are supported by simulation and show that this work improves the service quality by minimizing the handoff call dropping probability and maintaining the bandwidth utilization efficiently. Copyright © 2006 John Wiley & Sons, Ltd.     

基于无线局域网的实时视频传输与控制

针对无线局域网,研究实现了一种基于RTP/UDP/IP协议的端到端H.263实时视频传输系统.针对无线QoS机制系统应用了一种RTCP反馈友好码率控制算法TFRC,根据丢包率来动态调整发送码率实现拥塞控制.实验表明该传输方案能获得较好的视频传输质量.     

因特网IP QoS机制及其在UMTS网络中的应用

目前广泛使用的因特网协议(IP)已经成为所有数据业务的传输标准。传统的因特网业务策略(BE策略)不能满足新兴的实时业务的需求,因此有必要引入IPQoS机制来更有效的分配资源。由于第三代通信系统网UMTS需要同时支持非实时和实时业务,而且无线网的资源和设备比固定网昂贵得多,所以IPQoS机制对于无线网络尤为重要。文中首先研究因特网IPQoS机制,然后讨论如何将其应用到UMTS网络中。     

A QoS-aware routing mechanism for multi-channel multi-interface ad-hoc networks

To accommodate real-time multimedia application while satisfying application QoS requirements in a wireless ad-hoc network, we need QoS control mechanisms. In this paper, we propose a new routing mechanism to support real-time multimedia communication by efficiently utilize the limited wireless network capacity. Our mechanism considers a wireless ad-hoc network composed of nodes equipped with multiple network interfaces to each of which a different wireless channel can be assigned. By embedding information about channel usage in control messages of OLSRv2, each node obtains a view of topology and bandwidth information of the whole network. Based on the obtained information, a source node determines a logical path with the maximum available bandwidth to satisfy application QoS requirements. Through simulation experiments, we confirmed that our proposal effectively routed multimedia packets over a logical path avoiding congested links. As a result, the load on a network is well distributed and the network can accommodate more sessions than QOLSR. We also conducted practical experiments using wireless ad-hoc relay nodes with four network interfaces and verified the practicality of our proposal.     

基于SIP无线视频监控系统实时视频的实现

由于无线视频监控系统朝着网络化、一体化方向发展,提出了无线视频监控系统的实时视频的请求建立连接过程,采用简单性、扩展性较高的SIP作为呼叫连接控制协议,而在流媒体的传输过程中采用RTP/RTCP协议,增强实时视频灵活性,解决了实时视频对传输质量、传输速率等问题,并简要分析了客户端与服务器端的实时视频传输过程,结果显示视频图像清晰。     

Quality assurance of voice over WLANs (VoWLANs) with differentiated services

Several technical issues make commercial and large voice over wireless local area network (VoWLAN) services difficult to provide. The most challenging issue when voice over Internet Protocol (VoIP) services are ran over IEEE 802.11-based WLANs is the bandwidth inefficiency due to the considerable overhead associated with WLAN packet transmission. In this work, we propose a session-based quality-of-service management architecture (SQoSMA) to overcome the low number of VoIP calls in IEEE 802.11 Wireless LANs and the negative effect of new call addition when the WLAN reaches its capacity. The SQoSMA combines data and control planes to detect VoWLAN QoS degradations and performs either an adaptive audio codec switching or a call stopping to fix VoWLAN issues in a differentiated services manner. In addition, our solution deals with user sessions information, by considering user priority (from its agreement) to guarantee a certain level of its multimedia applications. Performance evaluation using a real test-bed shows that call codec change and call stopping techniques can easily assure high-priority calls with acceptable call blocking probability.     

Application-Level QoS Negotiation and Signaling for Advanced Multimedia Services in the IMS

The IP multimedia subsystem (IMS) has been recognized as a reference next-generation network architecture for offering multimedia services over an Internet Protocol (IP)-based infrastructure. One of the key benefits of the IMS is efficient and flexible introduction of new services and access to third-party application providers, thanks to standard interfaces and standardized service capabilities. To support novel media-rich applications across a wide range of user devices and access networks, IMS must support negotiable quality of service (QoS) for IP multimedia sessions. In this article, we describe the application-level QoS signaling as specified by the 3GPP and propose some enhancements based on advanced QoS parameter matching and optimization functionality to be included along the signaling path. We outline various signaling flow scenarios and discuss them in the context of a case study involving an IMS-supported 3D virtual environment, featuring a treasure-hunt-like game.     

DiffServ resource allocation for fast handoff in wireless mobileInternet

The next-generation wireless networks are evolving toward a versatile IP-based network that can provide various real-time multimedia services to mobile users. Two major challenges in establishing such a wireless mobile Internet are support of fast handoff and provision of quality of service (QoS) over IP-based wireless access networks. In this article, a DiffServ resource allocation architecture is proposed for the evolving wireless mobile Internet. The registration-domain-based scheme supports fast handoff by significantly reducing mobility management signaling. The registration domain is integrated with the DiffServ mechanism and provisions QoS guarantee for each service class by domain-based admission control. Furthermore, an adaptive assured service is presented for the stream class of traffic, where resource allocation is adjusted according to the network condition in order to minimize handoff call dropping and new call blocking probabilities     

QoS Provisioning in Wireless IP Networks

This paper deals with quality of service (QoS) provision in wireless IP networks. QoS provision is particularly challenging in wireless networks, where network resources are generally limited, variable over time and shared. In the design of possible measures to assure QoS one should consider that standardization is well established for the network layer Internet Protocol and for many underlying technologies of frequent use (e.g. IEEE 802.11, BLUETOOTH or HIPERLAN II). Therefore, as far as research on QoS is concerned, there is little room in both the IP and the link-layers for improved IP over wireless interfaces. In this paper we illustrate a solution in which an intermediate Wireless Adaptation Layer (WAL) is transparently interposed between the IP layer and specific link-layer technologies as a solution to provide QoS. The WAL addresses two main issues: (i) compensation for channel impairments in different platforms in order to enhance wireless channel reliability and (ii) implementation of traffic control and packet scheduling mechanisms to satisfy bandwidth and delay requirements, as well as to enforce a general principle of fairness among the IP associations contending for network resources and achieve optimal exploitation of transmission capacity. The WAL consists of a set of modules, each one in charge of a specific task, which can be enabled or disabled depending on the specific network environment. The novelty of the WAL approach is its capability of adapting itself to different wireless interfaces selecting performance enhancing modules for specific networks. This requires to modify the standard TCP/IP protocol stack by introducing an intermediate layer between the IP layer and the Data Link layer, with performance enhancement purposes. This paper focuses on two modules in particular, namely a traffic control module, which is in charge of performing congestion control and channel state dependent scheduling (CSD) packet scheduling, and a forward error correction (FEC) module, which compensates for channel impairments. This paper presents the proposed architecture provided with these modules and reports some measurements and simulations highlighting benefits resulting from the use of such modules.     

ECBRP: An Efficient Cluster-Based Routing Protocol for Real-Time Multimedia Streaming in MANETs

The dynamic characteristics of wireless networks and stringent QoS requirements of multimedia applications identify significant challenges for providing QoS guarantees for real-time multimedia streaming in such wireless environment. QoS routing protocols can decisively contribute to the QoS provision of network systems. This paper proposes an efficient cluster-based routing protocol (ECBRP) for real-time multimedia streaming in mobile ad hoc networks. First, to improve the stability of clusterheads, we introduce a new algorithm of cluster formation, in consideration of the node mobility and connectivity. Second, a link-broken detection mechanism is designed, which is able to distinguish whether packet loss is due to mobility or congestion, and to make proper reaction. This mechanism contributes to reduce route overhead, and to increase the decodable ratio of video frame at the application layer as well. Third, the routing protocol is enhanced via an adaptive packet salvage strategy, in order to alleviate the congestion in consideration of the characteristics of multimedia traffic. Our simulation experiment results demonstrate that the ECBRP leads to more stable cluster formation than the CBRP, and 80% decreases in the frequency of clusterhead changes against CBRP. As a result, the quality of real-time multimedia streaming is improved significantly, in terms of decodable frame ratio, delay and delay jitter, etc.     

H.323-SIP信令网关的实现

H.323和会话初始化协议（SIP）都是基于IP网络的多媒体通信协议,采用实时传送协议（RTP）传送实时的音频视频传输,二者之间可以通过设置信令网关互通,完成信令消息的翻译和转换功能。要实现H.323与SIP网络之间的互通,必须解决地址格式的转换、消息映射和终端能力协商等方面的问题。H.323-SIP信令网关在H.323和SIP互通中占有得要的地位,它能够使位于不同网络的用户直接和对端进行媒体交换,实现点到点、点到多点之间的通信。H.323-SIP信令网关的功能会越来越完善,也会越来越适应用户更加广泛的业务需要。     

QoS Provisioning in Wireless/Mobile Multimedia Networks Using an Adaptive Framework

Recently there is a growing interest in the adaptive multimedia networking where the bandwidth of an ongoing multimedia call can be dynamically adjusted. In the wireless/mobile multimedia networks using the adaptive framework, the existing QoS provisioning focused on the call blocking probability and the forced termination probability should be modified. We, therefore, redefine a QoS parameter – the cell overload probability – from the viewpoint of the adaptive multimedia networking. Then, we propose a distributed call admission control (CAC) algorithm that guarantees the upper bound of the cell overload probability. Also, a bandwidth adaptation algorithm which seeks to minimize the cell overload probability is also presented. Simulation experiments are carried out to verify the performance of the proposed CAC algorithm. Furthermore, the performance of the adaptive wireless/mobile network is compared to that of the existing non-adaptive wireless/mobile networks. As a further step in QoS provisioning, we propose another QoS parameter, the degradation period ratio, and discuss analytically how the CAC algorithm guarantees the upper bound of the degradation period ratio.     

Perceptive admission control for wireless network quality of service

As wireless networks become more widely used, there is a growing need to support advanced services, such as multimedia streaming and voice over IP. Traditional approaches to guarantee quality of service (QoS) work well only with predictable channel and network access. In wireless mobile networks, where conditions dynamically change as nodes move about the network, a stateless, high level approach is required. Since shared wireless resources are easily over-utilized, the load in the network must be controlled so that an acceptable QoS for real-time applications can be maintained. If minimum real-time requirements are not met, these unusable packets waste scarce bandwidth and hinder other traffic, compounding the problem. To enable high QoS for all admitted traffic, we propose the Perceptive Admission Control (PAC) protocol. PAC monitors the wireless channel and dynamically adapts admission control decisions to enable high network utilization while preventing congestion. Through discussion, simulations and testbed experiments, we demonstrate that PAC ensures low packet loss and delay for all admitted flows.     

Optimal network selection in heterogeneous wireless multimedia networks

The complementary characteristics of different wireless networks make it attractive to integrate a wide range of radio access technologies. Most of previous work on integrating heterogeneous wireless networks concentrates on network layer quality of service (QoS), such as blocking probability and utilization, as design criteria. However, from a user’s point of view, application layer QoS, such as multimedia distortion, is an important issue. In this paper, we propose an optimal distributed network selection scheme in heterogeneous wireless networks considering multimedia application layer QoS. Specifically, we formulate the integrated network as a restless bandit system. With this stochastic optimization formulation, the optimal network selection policy is indexable, meaning that the network with the lowest index should be selected. The proposed scheme can be applicable to both tight coupling and loose coupling scenarios in the integration of heterogeneous wireless networks. Simulation results are presented to illustrate the performance of the proposed scheme.