Multimedia synchronization protocols for broadband integratedservices

Protocols to provide synchronization of data elements with arbitrary temporal relationships of both stream and non-stream broadband traffic types are proposed. It is specified that the provision of a synchronization function be performed within a packet switched network, and, accordingly, a two-level communication architecture is presented. The lower level, called the network synchronization protocol (NSP), provides the ability to establish and maintain individual connections with specified synchronization characteristics. The upper level, the application synchronization protocol (ASP), supports an integrated synchronization service for multimedia applications. The ASP identifies the temporal relationships among an application's data objects and manages the synchronization of arriving data for playout. The proposed NSP and ASP are mapped to the session and application layers of the open-systems-interconnection (OSI) reference model, respectively     

多媒体远程教学系统实时传输解决方案

有效的传输机制是保障多媒体远程教学系统这样的分布式多媒体系统应用需求的关键。针对分布式多媒体应用,结合流媒体的传输控制协议,提出了一种远程教学系统实时传输解决方案:通过RTSP(实时流传输协议)建立和控制多媒体会话,RTP(实时传输协议)承载实时信息的传输,RTCP(实时传输控制协议)监控数据的传输状态,RSVP(资源预留协议)预约传输所需的Internet资源保证服务质量。该传输方案不仅适用于远程教学系统,也适用于其他基于Internet的实时多媒体应用系统。     

Media Synchronization and QoS Packet Scheduling Algorithms for Wireless Systems

Wireless multimedia synchronization is concerned with distributed multimedia packets such as video, audio, text and graphics being played-out onto the mobile clients via a base station (BS) that services the mobile client with the multimedia packets. Our focus is on improving the Quality of Service (QoS) of the mobile client's on-time-arrival of distributed multimedia packets through network multimedia synchronization. We describe a media synchronization scheme for wireless networks, and we investigate the multimedia packet scheduling algorithms at the base station to accomplish our goal. In this paper, we extend the media synchronization algorithm by investigating four packet scheduling algorithms: First-In-First-Out (FIFO), Highest-Priority-First (PQ), Weighted Fair-Queuing (WFQ) and Round-Robin (RR). We analyze the effect of the four packet scheduling algorithms in terms of multimedia packet delivery time and the delay between concurrent multimedia data streams. We show that the play-out of multimedia units on the mobile clients by the base station plays an important role in enhancing the mobile client's quality of service in terms of intra-stream synchronization and inter-stream synchronization. Our results show that the Round-Robin (RR) packet scheduling algorithm is, by far, the best of the four packet scheduling algorithms in terms of mobile client buffer usage. We analyze the four packet scheduling algorithms and make a correlation between play-out of multimedia packets, by the base station, onto the mobile clients and wireless network multimedia synchronization. We clarify the meaning of buffer usage, buffer overflow, buffer underflow, message complexity and multimedia packet delay in terms of synchronization between distributed multimedia servers, base stations and mobile clients.     

Efficient Media Synchronization Mechanism for SVC Video Transport over IP Networks

The scalable extension of H.264, known as scalable video coding (SVC) has been the main focus of the Joint Video Team's work and was finalized at the end of 2007. Synchronization between media is an important aspect in the design of a scalable video streaming system. This paper proposes an efficient media synchronization mechanism for SVC video transport over IP networks. To support synchronization between video and audio bitstreams transported over IP networks, a real‐time transport protocol/RTP control protocol (RTP/RTCP) suite is usually employed. To provide an efficient mechanism for media synchronization between SVC video and audio, we suggest an efficient RTP packetization mode for inter‐layer synchronization within SVC video and propose a computationally efficient RTCP packet processing method for inter‐media synchronization. By adopting the computationally simple RTCP packet processing, we do not need to process every RTCP sender report packet for inter‐media synchronization. We demonstrate the effectiveness of the proposed mechanism by comparing its performance with that of the conventional method.     

Towards a better-than-best-effort forwarding service for multimediaflows

A salient requirement of interactive multimedia applications is that they transmit data continuously at uniform rates with minimum possible end-to-end delay. The majority of these applications do not require hard and fast guarantees of network performance, but the current best-effort forwarding model of the Internet is frequently insufficient for realizing these requirements. Worse still, the requirement of uniform-rate transmission puts many multimedia applications at odds with current and proposed Internet network management practices that assume or require TCP-like reactions to packet loss. We are investigating router-based active queue management, specifically the use of queue occupancy thresholds to isolate TCP flows and to provide a better-than-best-effort forwarding service for flows in need of uniform-rate transmissions. Our current scheme, class-based thresholds (CBT), relies on a packet marking mechanism such as those proposed for realizing differentiated services on the Internet. CBT, when combined with existing active router queue management schemes such as random early detection (RED), provides a performance for TCP that approximates that achievable under a packet scheduling scheme and acceptable performance for multimedia flows. CBT is a simple and efficient mechanism with implementation complexity and run-time overhead comparable to that of RED     

MultiSync: a synchronization model for multimedia systems

Synchronization among various media sources is one of the most important issues in multimedia communications and various audio/video (A/V) applications. For continuous playback (such as lip synchronization) under a time-sharing multiprocessing operating system (such as UNIX), the synchronization quality of traditional synchronization mechanisms employed on single processes may vary according to the workload of the system. When the system encounters an overload situation, the synchronization usually fails and, even worse, results in two fatal defects in human perception: the audio discontinuity (audio break) and the out-of-synchronization (synchronization anomaly). In order to overcome these problems, a novel media synchronization model employed on multiple processes (or threads) in a multiprocessing environment is proposed. The problem of asynchronism due to system overload is solved by assigning a higher priority to more important media and adopting a delay-or-drop policy to treat the lower priority ones. Some experimental results are presented to show the effectiveness of the proposed model and the implementation mechanisms under a UNIX, X-Windows environment. On the basis of the proposed model, a continuous media playback (CMP) module, which acted as the key component of some popular multimedia systems such as multimedia authoring system, multimedia E-mail system, multimedia bulletin board system (BBS), and video-on-demand (VoD) System, was implemented     

Synchronization of multimedia data for a multimedia news-on-demandapplication

We present a complete software control architecture for synchronizing multiple data streams generated from distributed media-storing database servers without the use of a global clock. Independent network connections are set up to remote workstations for multimedia presentations. Based on the document scenario and traffic predictions, stream delivery scheduling is performed in a centralized manner. Certain compensation mechanisms at the receiver are also necessary due to the presence of random network delays. A stream synchronization protocol (SSP) allows for synchronization recovery, ensuring a high quality multimedia display at the receiver. SSP uses synchronization quality of service parameters to guarantee the simultaneous delivery of the different types of data streams. In the proposed architecture, a priority-based synchronization control mechanism for MPEG-2 coded data streams is also provided. A performance modeling of the SSP is presented using the DSPN models. Relevant results such as the effect of the SSP, the number of synchronization errors, etc., are obtained     

浅析分组传送网和传统MSTP网络混合组网

随着传统语音业务比重越来越弱化,数据业务的需求越来越多,各运营商网络建设都向分组网偏移。由于目前网络基数已经很庞大,部署分组网将是一个缓慢的过程,这就可能出现分组网与现有MSTP网络并存的情况,并有可能出现混合组网。分析了分组网和传统MSTP网络各自的特性,并基于各自不同的原理,对两种网络的融合方式、对接提出了不同的思路,并讨论其优劣性。     

Multimedia data delivery using network delay modeling

Most multimedia systems are by nature distributed. Stored digital media applications such as video-on-demand involve many separate clients and servers; communication applications such as videoconferencing involve users in distinct geographic locations. The retrieval of multimedia data for these applications across computer networks must be done in a timely fashion to accommodate end-to-end delay constraints, buffer space limitations, and inter-media synchronization. In this paper, we present a mechanism called thelimited a priori (LAP) scheduler which manages the retrieval of distributed multimedia data using network delay modeling. The LAP scheduler determines network load changes and estimates packet delay using a dynamic filtering algorithm. We show the bounds of accuracy for this technique and describe its suitability with respect to digital media traffic across a general-purpose network.Portions of this paper were presented at the 18th Annual Conference on Local Computer Networks, Minneapolis, MN, September 1993. This work is supported in part by the National Science Foundation under Grant No. IRI-9211165.     

Serving radio network controller relocation for UMTS all-IP network

To support real-time multimedia services in UMTS all-IP network, Third-Generation Partnership Project TR 25.936 proposed two approaches to support real-time serving radio network controller (SRNC) switching, which require packet duplication during SRNC relocation. These approaches significantly consume extra system resources. This paper proposes the fast SRNC relocation (FSR) approach that does not duplicate packets. In FSR, a packet buffering mechanism is implemented to avoid packet loss at the target RNC. We propose an analytic model to investigate the performance of FSR. The numerical results show that packet loss at the source RNC can be ignored. Furthermore, the expected number of packets buffered at the target RNC is small, which does not prolong packet delay.     

IEEE1588协议及其在路由交换平台中的实现技术

下一代网络（NGN）是一种融合了IP技术和多媒体通信技术的全新网络,然而当涉及到传统TDM业务应用及需要进行时钟同步分配时,基于IP技术的全新网络则需要具备完善的时钟同步能力来满足相关业务的同步需求。IEEE1588协议标准的出现正好解决了在新一代路由交换平台中的时钟同步问题。这里分析了IEEE1588协议的偏移测量和延时测量时钟同步过程,并给出了IEEE1588协议在路由交换平台中的具体实现过程。     

基于漏窗机制实时自适应多媒体同步

本文分析了实时多媒体同步问题以及相关研究,提出了一个基于漏窗机制实时自适应多媒体同步方案,该方案具有算法简单的特点,并且具有良好的自适应能力,可以适应各种网络变化,各种延迟特性,同样也适用于各种媒体类型,完全可以满足实时多媒体同步的需求。     

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.     

Cross-layer cut-through switching mechanism for IEEE 802.16d/e wireless networks

This paper proposes a fast cross-layer cut-through switching mechanism (CCSM) for supporting media access control (MAC) layer packet switching in IEEE 802.16-based broadband wireless access (BWA) networks. The local traffic, which means subscriber stations (SSs) communicating with each other within the cell, can be switched via the MAC layer without involving the network layer. The average access delay of request from SSs is studied and analyzed in this paper. Finally, the simulation and numerical results show that the performance of CCSM is superior to that of the legacy IEEE 802.16d/e protocol.     

Performance analysis of multiple access protocols for multimediasatellite networks

In this paper, several TDMA-based packet multiple access protocols are studied and evaluated in the geostationary satellite environment. The distributed queueing random access protocol, DQRAP, originally proposed for HFC networks is adapted to the satellite environment. Another protocol, the announced retransmission random access protocol, ARRA, proposed for wireless networks is also studied. Both protocols are modeled and simulated in a VSAT network context. We then propose a new protocol which combines the advantages of both studied schemes and is more adapted to interactive multimedia applications over satellite uplinks. The generalized retransmission announcement protocol, GRAP, regroups the immediate access by contention at low loads, and the reservation access. At higher loads, to achieve a better channel efficiency. An analytical model is proposed to calculate the channel throughput obtained by GRAP under different loading conditions. Simulation results illustrate an improved throughput/delay characteristics and a higher protocol stability compared to both DQRAP and ARRA. Enhanced versions of the protocol are also proposed and evaluated to further improve its efficiency, with reasonable additional complexity     

Design and analysis of cellular mobile communications system basedon DQRAP/CDMA MAC protocol

Future third-generation mobile communication systems will need multi-access control (MAC) protocols suitable for multimedia code division multiple access (CDMA) radio communications. Distributed queueing random access protocol (DQRAP)/CDMA is a general purpose MAC protocol oriented to the CDMA environment. Analytical model expressions and computer simulations have shown its capacity to achieve near-optimum performance under heterogeneous traffic scenarios in a unicellular environment. A cellular environment has been designed to verify that DQPAP/CDMA maintains its near-optimum performance in a packet switched mobile communication system. A new handover technique based on the protocol is proposed to further improve the system performance     

Multipath Security Aware Routing Protocol for MANET Based on Trust Enhanced Cluster Mechanism for Lossless Multimedia Data Transfer

The dynamic nature of mobile nodes of ad hoc network is mostly affected by security problems which reduce data forwarding rate in multimedia sources. Due to the rapid growth of wireless applications, the different multitalented routing protocols are proposed in recent years. But the recent protocols are not efficient for multimedia applications, till now, specific security aware routing protocols are not proposed for multimedia data transfers. In this paper, we proposed trust enhanced cluster based multipath routing (TECM) algorithm. We use energy efficient PSO algorithm used to create cluster formation and cluster head, super cluster head are selected from trust values, which compute form proposed TECM algorithm. The multi trust factors are used for trust computation, such as frame/packet loss ratio, frame/packet forward energy, frame/packet receiving energy, routing overhead, received signal strength, frame/packet forward rate, average forward delay and protocol deviation flag. We then combine proposed TECM algorithm with standard multipath OLSR protocol (TECM-OLSR) to analyze the performance of proposed algorithm. The simulated results show that proposed TECM-OLSR protocol is very effective in terms of loss and delivery rate, delay, routing overhead and network lifetime compare to FPNT-OLSR.     

车用自组网媒体访问控制机制改进

为提升车用自组网传输音频、视频的服务质量,对基于IEEE802.11p的车用无线接入技术MAC机制进行改进,提出竞争窗口自适应EDCA机制。仿真实验表明,竞争窗口自适应EDCA机制有效地降低了车用自组网中音频、视频流的传输时延、时延抖动和丢包率,保证了车用自组网传输VoIP、视频会议、音视频流媒体等多媒体业务的服务质量。     

MSTP体系结构研究

随着IP数据、话音和图像等多种业务传输需求的不断提高,现有以承载话音为主要业务的城域网在容量以及接口能力上都已经无法满足业务传输与汇聚的要求。以SDH为基础的多业务传输平台能够克服传统城域网的缺点,可以更有效地支持分组数据业务,并有助于实现从电路交换网向分组网的过度。介绍了MSTP的分类方法,给出了MSTP的分层结构,并对各功能模块的需求和特点进行了详细分析。     

ATM-based transport architecture for multiservices wirelesspersonal communication networks

This paper presents an ATM-based transport architecture for next-generation multiservices personal communication networks (PCN). Such “multimedia capable” integrated services wireless networks are motivated by an anticipated demand for wireless extensions to future broadband networks. An ATM compatible wireless network concept capable of supporting a mix of broadband ISDN services including constant bit-rate (CBR), variable bit-rate (VBR), and packet data transport is explored from an architectural viewpoint. The proposed system uses a hierarchical ATM switching network for interconnection of PCN microcells, each of which is serviced by high-speed, shared-access radio links based on ATM-compatible cell, relay principles. Design issues related to the physical (modulation), media access control (MAC), and data-link layers of the ATM-based radio link are discussed, and preliminary technical approaches are identified in each case. An example multiservice dynamic reservation (MDR) TDMA media access protocol is then considered in further detail, and simulation results are presented for an example voice/data scenario with a proportion of time-critical (i.e., multimedia) packet data. Time-of-expiry (TOE) based queue service disciplines are also investigated as a mechanism for improving the quality-of-service (QoS) in this scenario