BGP Blackholing is an effective mitigation solution for networks to counter the frequent Distributed Denial of Service (DDoS) attacks. It enables to drop all network traffic that is directed towards a particular victim prefix under DDoS attack, ideally, as close to the source as possible. Despite its huge importance in the Internet, there is no tool available for the real-time visualization of BGP Blackholing activity. Visualization is one of the most powerful techniques for network operators to monitor network activity. From discovering successful network topology to expose anomalous behaviors in networks, easy-to-use visualizations are powerful weapons to capture important patterns on the Internet traffic[1, 5]. In this work, we propose a first real-time BGP Blackholing Visibility Tool (named as SmartX BGP-BVT) to detect and visualize community based BGP Blackholing on live BGP data. This tool will be helpful for network operators and researchers interested in BGP Blackholing service and DDoS mitigation in the Internet.

Accurate Internet traffic predictions can provide support to network operators for applications such as traffic engineering, bandwidth allocation, anomaly detection, etc. We apply and compare different forecasting techniques (traditional and machine learning-based techniques) on real BGP data that is collected from two well-known Internet exchange points (IXPs) to derive BGP future volume-based predictions. Our experimental evaluation shows that multivariate Bayesian Ridge outperforms all other forecasting techniques we consider. Through univariate LSTM, we are able to predict new BGP volume-based features. Furthermore, to study the impact of dataset size on BGP forecasting, we perform experiments on three BGP dataset sizes, i.e., Short (one-month), Medium (three-months), and Long (five-months) Periods. Our results show that the Short-Period BGP dataset seems to be sufficient for getting accurate predictions. We also present a use case study (forecast Google Leak anomaly) that supports our experimental evaluations. We provide our collected BGP datasets publically which will be helpful to perform further research experiments and analysis regarding BGP traffic predictions.

Forecasting network traffic is a challenging task for better network management. In this poster, we present a Border Gateway Protocol (BGP) traffic volume prediction framework that uses real BGP data from two famous Internet exchange points (IXPs) to train the LSTM network and generate future volume-based predictions. Our experimental evaluation shows that LSTM can indeed be used to predict BGP traffic volume with a very low prediction errors.

In the cloud field, there is an increasing demand for globalized services and corresponding execution environments that overcome local limitations and selectively utilize optimal resources. Utilizing multi-cloud deployments and operations rather than using a single cloud is an effective way to satisfy the increasing demand. In particular, we need to provide cloud-native environment to organically support services based on a microservices architecture. In this paper, we propose a cloud-native workload profiling system with Kubernetes-orchestrated multi-cluster configuration. The contributions of this paper are as follows. (i) We design the operating software over multiple cloud-native cluster to select optimal resources by monitoring. (ii) For operating the multiple clusters through the design, we define and design specific general service workloads. Also, we implement the workloads in application software (iii) To seek optimal resources, we deployed the general workloads and monitored resource usage repeatedly in detail. We calculate resource variation in comparison with initial resource usage and average resource usage after deploying the service workloads. Also, we analyze the resource monitoring result. We expect this methodology can find proper resources for service workload types.

The ICT infrastructure transformations for integrating cloud, software-defined networking (SDN) and network function virtualization (NFV) are emerging extensively to satisfy dynamic user requirements on diversified functionalities over large-scale resource pools. These infrastructure transformations are causing new sets of operational problems for operators, as a consequence of inadequate end-to-end visibility and visualization support about underlay infrastructure. Particularly, for multi-tenant/multisite underlay infrastructure, resource monitoring and associated flow identification are major challenges. To address these operational challenges, precise visibility monitoring and interactive visualizations are essential. Therefore, in this paper, by realizing an interactive visualization of multi-layer (i.e. resource-layer and flow-layer) visibility, we address the real-world operational issues of experimental SDN-enabled distributed clouds, denoted as OF@TEIN and OF@KOREN Playgrounds. By leveraging ‘SmartX MultiView Visibility Framework’, we first strive to enable visibility collection and validation. We then link it with subsequent visibility integration capability empowered by DataLake-based visibility storage. Additionally, by taking the example of OF@TEIN and OF@KOREN Playgrounds, we present verification results on prototype implementations. The results justify the proposed approach for timely unveiling critical operational issues by efficaciously leveraging the multi-layer visibility.

The megatrends and Industry 4.0 in ICT (Information Communication & Technology) are concentrated in IoT (Internet of Things), BigData, CPS (Cyber Physical System), and AI (Artificial Intelligence). These megatrends do not operate independently, and mass storage technology is essential as large computing technology is needed in the background to support them. In order to evaluate the performance of high-capacity storage based on open source Ceph, we carry out the network performance test of Abyss storage with domestic and overseas sites using KOREN (Korea Advanced Research Network). And storage media and network bonding are tested to evaluate the performance of the storage itself. Additionally, the security test is demonstrated by Cuckoo sandbox and Yara malware detection among Abyss storage cluster and oversea sites. Lastly, we have proposed the draft design of Data Lake framework in order to solve garbage dump problem.

We discuss the prototype implementation of IoT-Cloud services for efficient cooling management for small-size server room. We leverage open-source-based miniaturized playground for IoT-Cloud services and realize smart energy IoT-Cloud service based on the container-leveraged service orchestration with Docker containers and Docker Swarm. ln addition, we attempt to facilitate the automated and reliable operation of IoT-Cloud services with container-based orchestration.

In this paper, we describe the concept of utilizing packet tracing and collection for 3rd-party packet-precise inspection to assist the secure operation of SDN-enabled multisite cloud environment. The proposed packet tracing and collection is designed and prototyped by utilizing open-source IO Visor project's tools. To verify its feasibility for remotely distributed multiple boxes, we apply it to the multi-box playground with distributed SmartX server-switch boxes. By combining the packet tracing data collected from distributed sites at a centralized location, it is verified to be useful in supporting the malicious packet detection in our multi-site playground.

Visibility measurement and associated visualization are essential for monitoring the operation of emerging converged software-defined infrastructure (SDI). However, most of existing monitoring tools are not yet ready to cover systematically whole visibility workflow. Thus, in this paper, we present an integrated design for multi-layered visibility workflow, denoted as 'SmartX MultiView Visibility Framework'. The proposed open-source framework targets to streamline the visibility workflow for collection, validation, integration, storage, staging, and visualization. As a proof of concept, initial-stage implementation is deployed over OF@TEIN multisite SDN-Cloud playground to visualize resource-layer visibility with overlapping inter-connect flow paths.

Cloud computing paradigm has been driving the cloud-leveraged refactoring of existing ICT services including VoIP. In this paper, we design and prototype secured mobile VoIP services with open-source Asterisk PBX software by employing Docker lightweight virtualization for mobile devices. We also experimentally verify the quality of secured voice and the associated communication delay over distributed global connectivity environment.

The VoIP technology is being revisited to support diversified services and mobile computing has been changing to mobile cloud computing paradigm. In this paper, we design and implement cloud-leveraged secured voice services, enabled by the combination of open-source software PBX (private branch exchange), shared hardware platform, and mobile devices. We also test and verify the end-to-end delay of the secured voice communication. In addition, we propose a preliminary idea to improve the portability by adopting Docker-based containers.

The focusing concept of Internet of Things (IoT) is quickly changing from inter-connecting uniquely-identifiable embedded devices to 'things as a service'. Thus, diverse IoT services are being incubated by software-driven frameworks and shared/virtual/programmable resources. As a beginning step, in this paper, we discuss how to prepare and inter-connect hyper-converged SmartX Boxes so that it can play the miniaturized playground for IoT-Cloud service realization.

In OF@TEIN (OpenFlow @ Trans-Eurasian Information Network) SDN-enabled distributed cloud testbed, the resource-level visibility is extremely critical to ensure high availability of Physical+Virtual resources and identifying key performance bottlenecks which can significantly degrade testbed operation. Among multiple visibility levels for resources, flows and services, resource-level visibility is becoming crucial. To address resource-visibility challenges, we setup SmartX Visibility Center as a component solution of SmartX Operation Tower by integrating resource-level visibility tools. In this paper, we present our idea of resource-level visibility with specific focus on performance data collection, distributed resources monitoring, events alerts generation and visualization of gathered testbed operation data over large-scale networked tiled displays.

In a WiFi network with mobile nodes, it is quite a challenging task to provide P2P (peer-to-peer) VoD (video-on-demand) service while maintaining high aggregated throughput of network. MOVi (mobile opportunistic video-on-demand) is an early trial for mobile P2P VoD service where a centralized coordinator schedules segment-based transmissions (i.e., segment scheduling) for mobile nodes based on the opportunity, which reflects the pairing chances among nodes. Since MOVi scheduling depends on this fluctuating-over-time opportunity, mobile nodes may not play the video contents smoothly in case of temporal opportunity shortage. To mitigate this, in this paper, we propose to extend MOVi, MOVi+, by employing special peer nodes with segment caching capability and applying pre-diffusion-aware scheduling. With prototyped MOVi+ implementation, we validate the improved performance of MOVi+ in terms of segment-missing ratio, opportunity usage, segment distribution completion time, and the number of simultaneous playout mobile nodes with acceptable playout continuity (i.e., over 95%).

This demonstration introduces our on-going work called MOVi+ that attempts to improve the video delivery performance of MOVi (Mobile Opportunistic Video-on-demand) by scheduling segments from the whole node perspective. By deploying segment based caching-enabled switch, MOVi+ increases the opportunity which reflects the pairing chances among nodes. MOVi+ also evaluates the merit of pre-diffusion-aware scheduling to utilize the time varying opportunity as much as possible. We shows that MOVi+ can eventually service more number of mobile nodes than MOVi with lower segment missing ratio, better opportunity usage, and playout continuity.

Near Field Communication (NFC) is an integration of contact less smart card communication technology into mobile devices, such as smart phones. NFC technology provides three operating modes and each mode differs from each other in terms of communication and data processing model. We design the NFC based Micro-payment system and security functions for trade safety. The sales of small retail trade and traditional markets are on a decreasing trend. In order to solve this problem, we propose the Micro-payment model based on NFC, tokenization technique to support the privacy of Micro-payment, and MD authentication technique for fasten Micro-payment in aspect of IT.

As the heterogeneity of end devices and their networked applications grows, networking challenges should be coped with new features to meet the various requirements of different application traffics. In this paper, based on the emerging SDN (Software-Defined Networking) paradigm, we introduce NetOpen networking services that attempt to support configurable networking with programmable networking switch nodes. Especially, we prototype a NetOpen switch node to support customized in-network processing and verify its programmability and in-network processing performance for multiple flows.

In this demonstration, we show our ongoing work called MOVi+ that extends MOVi (Mobile Opportunistic Video-on-demand) for improving the scalability in terms of number of mobile clients. By exploiting sustainable segment-based caching-enabled switch, MOVi+ increases the opportunity of peer-to-peer connectivity, which can eventually service more number of mobile clients than original MOVi with shorter startup time, better video quality, and playout continuity.

In this work-in-progress paper, we present a network coding scheme based on random linear coding (RLC), named cumulative RLC. Cumulative RLC applies RLC in a cumulative manner such that the end-to-end delay is minimized and it can be used for real-time one-to-many video distribution over wireless networks. The proposed cumulative RLC is well-suited especially for media applications with low-latency constraints such as real-time resilient video distribution for interactive service scenarios. The key property of cumulative RLC is reducing the waiting delay during the RLC encoding procedure. To verify the efficiency of the proposed cumulative RLC, we implement a real-time resilient video distribution system employing cumulative RLC over a IEEE 802.11-based wireless network. Experimental results show that in a typical setting, cumulative RLC yields reducing RLC encoding delay so it can help efficient real-time video distribution.

In spite of its success, BitTorrent is facing a content unavailability problem where peers can not finish their content download due to an absence of seeders. Although some approaches have been proposed to improve the content availability, the prior conditions such as a downloaded content and a set of related contents to be bundled limit their application scopes. Our main contribution is a multi-swarm collaboration to improve the content availability of the BitTorrent-like systems without the content-related limitations. The multi-swarm collaboration enables the collaborating swarms cache some chunks that are likely to be unavailable in near future of each other when the seeders are online and share the cached chunks when the content unavailability happens. Our approach enables any swarms that have appropriate amount of resources to collaborate with each other for the improved content availability. Through simulations, we show that the multi-swarm collaboration improves the peer performance as well as the content availability.

CPS (cyber-physical system) is a kind of system that exhibits a tight combination of system computation and physical elements. This tight integration of computing and physical elements is achievable depending on the performance of reliable data communication among CPS nodes. To support the domain-specific quality requirements of reliable real-time CPS networking, we should deploy a network-adaptive reliable data distribution scheme while paying extra attention to the incurring computational complexity of selected error control modules. In this paper, by taking the example of reliable real-time data distribution for UAV (unmanned aviation vehicle)-aided tactical networks, we discuss how to build a simulation environment that can evaluate the tradeoff of CPS networking performance and resource consumption (in terms of computing and networking resources). To validate the appropriateness of constructed simulation environment, we apply specific error control building blocks on the resource-constrained multi-level hierarchical UAV-aided tactical WiMax networks and evaluate their performance and resource consumption.

Future Internet Groups have been studying networking virtualization and computing virtualization for Test bed. Specially, GENI have adopted cloud computing as computing virtualization technique. In this paper, we describe the multistage anomaly detection scheme of anomaly and symptoms of DDoS attacks for secured cloud computing resource. This scheme is composed by 3 stages. Monitoring stage performs misuse detection by rules of attack patterns to well known DDoS attacks. Lightweight anomaly detection stage could classified volume data into large volume data and small volume data, and applied Bayesian methods to detect anomaly and symptoms of DDoS attack. And focused anomaly detection stage is performed to detect novel DDoS attacks by unsupervised learning algorithm.

Transmission scheduling to improve the capacity in wireless mesh networks (WMNs) is challenging. One promising candidate solution is backpressure scheduling, which was shown to provide throughput optimality and queue stability. Several recent practical systems that implement approximations of backpressure scheduling show performance benefits in WMNs that are based on IEEE 802.11, which is the most widely adopted MAC protocol. However, a detailed analysis of the queuing behavior is still missing for practical systems. In this paper, we experimentally show that backpressure scheduling over IEEE 802.11 presents significant queue instability due to the delayed interaction between MAC and network layers. To understand how and when queue instability occurs, we present a weighted backpressure scheduling (WBS), which uses two additional weight factors β and γ for the queue lengths of a node and its next-hop node when computing the backpressure value. By using WBS, we experimentally investigate the interactions between the components of backpressure scheduling implementations, i.e packet scheduling and link scheduling, and show that, usually, β >; γ leads to queue stabilization.

In this paper, we carry out a transparency (i.e., haptic realism) analysis and propose a haptic synchronization scheme for a force-reflecting teleoperation over time-varying network delay. Although previous researches proposed several haptic synchronization schemes to efficiently compensate for delay jitter, defining a proper synchronization delay for transparency improvement remains a challenge. Therefore, the transparency analysis quantifies the force feedback distortion caused by network delay and packet loss, and predicts the maximally allowable delay and loss for predefined transparency requirements. In order to minimize the force feedback distortion, the proposed haptic synchronization scheme controls the playout times of the delayed transmitted haptic events based on the transparency analysis. According to simulation and experimental results, the transparency analysis provides an acceptable quantification method about haptic interaction quality with respect to network delays and packet losses. It is also verified that the proposed scheme guarantees more transparent haptic interaction than existing synchronization schemes tailored for networked haptics.

The volume of P2P application traffic has increased so much that network-operator-driven traffic control techniques such as P4P for localizing the P2P traffic are recently proposed. The gist of the existing approach is that the network operators provide traffic and topology information as a guidance to peers so that P2P traffic will flow as the network operators intend, thus, realizing unilateral interaction from the network operators to the peers. Thus, this paper proposes bilateral cooperation between the network operators and the peers in P2P networks so that both parties may enjoy even further benefit in reducing cross-domain traffic and in optimizing content download time. In a nutshell, the peers provide peer-level network information to the network operators, which would otherwise require the network operators to perform costly flow-level analysis.We divide measurement work into two parts, letting the peers collect fine-grained traffic information and enabling the network operators to grasp macroscopic information in order to issue a useful guidance to peers including traffic bound missing in the existing work. Our simulation result shows that the bilateral cooperation reduces maximum link utilization of intra links by 34.41%, cross-network traffic by 23.02%, and content download time by 4.65% compared to the existing unilateral interaction.

Haptic-based NVEs (networked virtual environments) with CS (client/server) communication architectures support better consistency but induce larger end-to-end delays than those with P2P (peer-to-peer) communication architectures. Unfortunately, large delay severely deteriorates the transparency (i.e., reality) of haptic interaction. To improve the haptic interaction quality for haptic-based NVEs with CS communication architectures, in this paper the degradation of haptic interaction quality is analyzed according to network delays. Based on the analysis, the maximum allowable delay bound is predicted and unrealistic force feedback caused by the network delay is compensated. Experimental results confirm the proposed delay compensation scheme effectively improves haptic interaction quality with respect to network delays.

This paper analyzes TCP fairness in IEEE 802.11-based wireless mesh networks (WMNs). Fairness of bandwidth sharing is important for equitable network access by various users. With our Indoor WMN testbed, we observe that even though TCP flows have similar round trip time and loss probability, they encounter an unfairness problem. Especially, when TCP flows experience high TCP ACK losses (around 20%), unfairness happens in a relatively long period (1~100 seconds). We denote this phenomenon as a semi-long-term unfairness in the same path (SLU-SP) problem because it happens even when TCP flows share the same routing path. Additionally, we find out that the extended hidden terminal problem boosts up high TCP ACK losses and leads to a serious SLU-SP problem. Also, by carefully observing TCP dynamics through an active monitoring tool, PaPMo (packet-accurate protocol monitor), we find out that the cause of SLU-SP problem comes from the cumulative ACK mechanism of TCP. Cumulative ACK makes TCP flows tolerable to few ACK losses, but when the TCP flows have different sizes of congestion window, the TCP flows have unfair bandwidth sharing in the relatively long period.

With HPC (high performance computing) and scientific computing, it has been possible to produce a large amount of data. To easily analyze those data and collaborate on it, visualization in terms of ultra high resolution is desired. However, in order to combine a HPC system with a high resolution display system, high system resource is required. This paper shows an example of a system combining both a haptic system requiring high speed computing and a ultra high resolution display system. Thus, in this paper, we realize a networked haptic virtual environment system over ultra high resolution tiled display. With a networked haptic virtual environment supporting ultra high resolution display, users can feel sense of touch with haptic device and see visuals in ultra high resolution manner with network tiled display when users collaborate in shared virtual environment together. In addition, in this paper, a frame rate control scheme, making the networked haptic virtual environment system over ultra high resolution display stable, is proposed.

In this paper, we will study an broadcast protocol based on prioritization for delay-tolerant networks (DTNs). Considering the feature of the DTNs where there may be rare stable links, we propose an application-oriented broadcast (AOB) protocol built atop "bundle layer" in DTNs, which adjusts the forwarding probability for different types of traffic categories. As a result, AOB can provide differentiated services for prioritized data packets and simultaneously reduce the broadcast overhead. The performance of the proposed protocol is evaluated by extensive simulations for scenarios with not only random way point mobility model but also realistic mobility traces collected from cars moving in the city.

For the mobile users (MUs) installed with IEEE 802.11 wireless adapters, the default best-signal-strength-based access point (AP) selection is limited since it does not consider fairness and multi-rate issues. In this paper, we propose a distributed AP selection algorithm for newly joining MU and already associated MUs to be adaptive to their joining order and the dynamic wireless environment. We also prove the stability of proposed algorithm and analyze its complexity. Finally, with extensive evaluations, we show that proposed scheme can greatly improve the fairness without significantly sacrificing the efficiency.

To support diverse real-time applications, the next-generation Internet needs to provide end-to-end performance guarantee on a per-connection basis in the integrated service environment. In addition to the issues of how to allocate resources to satisfy the specific quality-of-service (QoS) requirements in a single switch, we also need to prevent the burst arrivals due to complex interaction of the packets from different sources within intermediate switches. To address this problem, a new frame-based packet scheduling scheme: 2-tier hierarchical frame (2tHF) queueing, is proposed and investigated in this paper. The key idea of 2tHF queueing is that packets to be served within a frame are evenly distributed over the multiple sub-frames and served in the order of time with other packets. Through numerical analysis, we have shown that when using varying number of sub-frames, the proposed 2tHF queueing not only provides the differentiated end-to-end delay bound, but also solves the coupling problem between the bandwidth allocation granularity and delay performance.

Wireless mesh networks (WMNs) are promising last-mile access network with large coverage and low deployment and operational cost. As the realization of WMNs becomes attainable, business opportunities have focused a number of applications such as multimedia streaming. However, in the case of multiple clients streaming, intrinsic nature of wireless medium sharing property results severe streaming bandwidth unfairness among the clients. To cope with this problem, this paper proposes an effective and practical distributed solution for fair bandwidth allocation among streaming clients in IEEE 802.11-based WMNs based on proportionally normalized congestion price. Especially, we approximate the congestion price from the intermediate streaming flow rate, and finally coordinate the resulting WMNs by taking the proportional normalization to the congestion price. As validation, ns-2 simulations are conducted and the results show that, under heavily congested streaming scenario, the proportional fairness is improved almost two times compared to the original IEEE 802.11 WMN.

Live streaming of multimedia content over the Internet is rapidly increasing, e.g. with the Web 2.0 and via peer-to-peer systems. While many homes today are equipped with sufficient bandwidth (DSL) to support high quality live streaming, the quality of experience of a user is dismal: the video resolution and size is small and the displaying is often interrupted. One of the key problems arises because TCP is used as the underlying transport layer protocol: packet loss, retransmissions and timeouts prevent TCP from delivering a timely, steady flow of packets and thus inhibits the provision of online multimedia content. This paper presents TCP-ROME, a novel transport-layer framework that allows an establishment of multiple TCP connections between one client with possibly multiple home addresses and multiple co-located or distributed servers. The download over multiple TCP connections increases the total throughput by aggregating the resources of multiple TCP connections. Moreover, TCP-ROME provides the means to overcome bandwidth fluctuations and network bottlenecks by dynamically coordinating the downloaded content from the different servers and by coordinating and adapting the streaming rate of the different connections to meet the bandwidth requirements of the video.

The multimedia-oriented collaboration is very important for knowledge workers who work together at a distance in pervasive computing environments. In this demonstration, we present a multi-agent-based toolkit to manage the coordination of multiple component services for multimedia-oriented collaboration environments. Based on a multi-agent system, the proposed toolkit is designed to support features such as node configuration for multiple devices, flexible operation of versatile collaboration tools, and session management for multi-party collaboration.

This paper introduces the software design for smart meeting space (SMeet) that enables users to collaborate with remote persons by using a wide range of devices embedded in networked meeting rooms. To enhance the quality of user experience in remote collaboration, SMeet is designed to satisfy several considerations from the integrated view of media, networking, display, and interaction services. We propose a software architecture of SMeet to flexibly associate component services to support high-quality media transport, high-resolution tiled display, and display interaction. We also present an approach to create composite services by combining component services. We explain key issues to dynamically build composite services and to integrate the service composition scheme with the proposed software architecture.

Providing on-demand video streaming services to large numbers of nodes in mobile wireless networks is a challenging task. We propose MOVi (mobile opportunistic video-on-demand), a mobile peer-to-peer video-on-demand application based on ubiquitous WiFi enabled devices such as smartphones and Ultra Mobile PCs. MOVi poses new challenges in on-demand video streaming caused by limited wireless communication range, user mobility and variable user population density. MOVi addresses these challenges by exploiting the opportunistic mix use of downlink and direct peer-to-peer communication to significantly improve achievable overall system throughput. Through simulation and actual implementation, we observed that MOVi is able to increase the number of supported concurrent users two fold compared with unicast based on-demand video streaming, as well as reducing video start-up delay by half. Significantly, these performance improvements can be achieved without increasing power consumption at the end device. Using MOVi, users not only share and distribute on-demand video such as YouTube clips for their own performance gain but also for the benefit of the collaborative group.

In this paper, we address access point (AP) selection for mobile nodes (MNs) in IEEE 802.11 wireless mesh network (WMN). Unlike the hotspot-based AP model, the service capacity of mesh AP in WMN is limited by the wireless connection to gateway AP and varies with the hop distance to the gateway AP. Using the multi-rate throughput model of (1), we formulate the problem of AP selection in WMN as a nonlinear optimiza- tion problem where the tradeoff between total throughput and fairness is achieved by applying a weighting parameter λ .W e evaluate the tradeoff over several WMN topologies, and show that the AP selection result from the proposed formulation can result in a reasonable tradeoff.

In multi-party collaborative environments, a group of users can share multiple media streams via IP multicasting. However, despite of the efficiency of IP multicast, it is not widely available and alternative application-layer multicast approaches are introduced. Application-layer multicast is advantageous, however, it incurs additional processing delays. In this paper, we present a new hybrid-style application-layer multicast solution that satisfies both network efficiency and easy deployment. We achieve this goal by connecting multicast islands through UDP tunnels employing UMTP (UDP multicast tunneling protocol). We also design a MPROBE protocol to remove multicast loop among multicast island in real Internet. We verify the feasibility of the proposed solution by implementing a prototype tool, AG Connector, that works on Access Grid multi-party collaborative environment.

To realize the real-time network-adaptive video streaming service over bandwidth-scarce and fluctuating WLAN, an effective channel monitoring scheme is required. In this paper, we propose a cross-layered network-adaptive streaming system where an intelligent intermediate sender with an embedded wireless access point estimates the time-varying channel condition (namely, Sender-side Cross-Layered Monitoring: CLMS). For a given monitoring interval, the CLMS estimates the available time resource for actual data transmission (i.e., channel time) by using observed MAC (medium access control) and PHY (physical) statistics. A streaming sender, which has an interface with the CLMS, then dynamically adjusts the scalability of video (e.g., temporal scalability) according to the estimated channel condition. Evaluation results over the IEEE 802.11a WLAN testbed is provided to verify the intended QoS (quality of service) adaptation capability. The experimental results show that the proposed system with the CLMS can effectively mitigate the quality degradation of video streaming caused by the fluctuations of time-varying wireless channel condition.

In this paper, a network adaptation scheme that involves a haptic event prioritization concept is presented. It classifies the haptic events and efficiently adapts the transmission and error rate according to the importance of the haptic events. According to the experimental results, the proposed scheme provides higher haptic interaction quality with a shorter playout delay and a lower transmission rate compared to existing networked haptic transport schemes.

Wireless networks are inherently vulnerable to loss from dynamic wireless channel. Particularly, wireless channel is affected by time-varying fading and interference, which leads to burst packet losses and thus causes severe quality degradation of video streaming. In order to alleviate the effect of losses in the transport layer, the error control schemes such as FEC (forward error correction), ARQ (automatic repeat request), and interleaving are essential. However, since each error control scheme shows different performance according to the transport application and the status of channel, a suitable scheme should be selected and applied. In this paper, we propose a network-adaptive selection of transport error control (NASTE). The proposed mechanism selects a suitable error control mode according to the status of channel and application. To verify the feasibility of the proposed NASTE, we have implemented it over a real IEEE 802.11g WLAN (wireless LAN) environment. The experimental results indicate that the proposed mechanism can enhance the overall transport performance by recovering packet losses.

Understanding the maximum end-to-end throughput available with any given wireless multi-hop network is an important, but very challenging task. It may be approximated up to certain degree for several simplified topologies such as chain, cross, and others. Typically, in case of chain-topology wireless multi-hop networks employing IEEE 802.11 DCF (distributed coordination function), the maximum throughput can be derived by calculating 1-hop average throughput considering the so-called bottleneck region. However, this approximation method that utilizes only the 1-hop average throughput is limited in terms of accuracy. Thus, in this paper, after verifying the limitation of 1-hop average approximation for the throughput calculation of chain-topology multi-hop wireless networks, a refined calculation scheme is proposed by additionally considering the deviation of throughput. Especially, we focus on the fact that the throughput deviation could lead to collisions, and the throughput deviation happens due to the summation of uniformly distributed backoff times of IEEE 802.11 DCF. With several ns-2-based network simulations, we verify the impact of selected temporal variations to the throughput calculation and thus validate the improved accuracy of proposed calculation scheme.

Teleconferencing solutions such as VoIP conferenc- ing systems employ acoustic echo cancellers to reduce echoes that are resulted from the coupling between loudspeaker and microphone. While echo cancellation in itself is a very delicate and complex task, requiring an exact modeling of the echo generating environment and a great amount of fine tuning in real-time, the task becomes even more complex in the case of a multi-party conferencing scenario. We propose a highly adaptive and efficient acoustic echo cancellation solution for multi-party VoIP conferencing systems. The proposed solution is based on a novel VoIP conferencing framework for high quality echo cancellation in a multi-party VoIP conferencing environment. All the building blocks of the proposed solution are completely specified and elaborated using a model VoIP conference call scenario. The proposed acoustic echo cancellation solution is simulated, tested, and verified using MATLAB for its high efficiency and quick adaptability. Results show that our proposed solution offers a unique conferencing experience for multi-party VoIP conferencing systems by delivering extremely good voice quality. I. INTRODUCTION In recent years the need for echo cancellation has appeared in a new area, namely hands-free telephony for which the demand is expected to increase enormously in the years to come. In hands-free telephony, there is an additional delay for the signals to travel via the telephone network. This can be quite long, depending on the telephone setup and connection, and the echoes may be noticeable. Therefore, in some way the echoes from the loudspeaker output that are picked up by the microphone must be removed from the microphone signal. A. Echo Cancellation in VoIP Echo cancellation is critical in achieving high quality voice transmissions over packet networks, which typically face transmission delays in excess of 30 to 40 ms. Such a long delay makes echo readily apparent to listeners, and must be eliminated for a viable telephony service. In this paper, we propose a highly adaptive and efficient acoustic echo cancellation solution for multi-party VoIP con- ferencing systems. The proposed solution is based on a novel VoIP conferencing framework for high quality echo cancel- lation in a multi-party VoIP conferencing environment. The proposed acoustic echo cancellation solution is implemented and its performance is evaluated using extensive simulations for its high efficiency and quick adaptability.

In this paper, we propose a practical media service framework to offer high-quality media service for multiparty collaborative environments. The proposed framework suggests an unified approach to employ various media formats, enable multicast-connectivity and display highresolution videos. It includes media controllers to enable DV (digital video) and HDV (high-definition DV) streaming, announce sessions, and guide the adaptive QoS control. It includes media connectors to enable using UDP tunneling to offer multicast-connectivity. Finally, it supports media producers/consumers to encode, transmit, decode, and render high-quality media. We explore the possibility of proposed framework by implementing a prototype and evaluating it over multicast-enabled network testbed.

As A/V (audio/video) services over IP networks become popular, there has been much interest in using home networking technology to provide QoS (quality of service)- guaranteed A/V services. For QoS-guaranteed A/V services, reservation or allocation schemes are usually used to control the demand of network bandwidth. However, these approaches have some limitations that discourage practical implementations. It requires that all applications participate in the reservation system. It can not manage actual network resource status, since it does not provide dynamic network resource monitoring scheme. To provide QoS-guaranteed A/V services, it is necessary to know actual network resource status, although all applications are not in the reservation system. We propose a middleware module that can improve existing reservation-style approaches, namely QoSAgent, based on standard home networking pro- tocol. Based on UPnP (universal plug and play) QoS services, the QoSAgent monitors network resource status and adapts its streaming services to varying network status. From the VideoLAN-based implementation, we confirm that the proposed approach can be used for QoS-guaranteed streaming services in home networks. I. INTRODUCTION With widespread Internet technologies, and the increasing number of users owning networking devices at home, home networks are becoming popular. The home network has been driven by home automation. Most of the applications that have been delivered to users are for home automation (11). Those applications include security, lighting, heating, cooking, washing appliances, etc. However, most recently, the effort on the development of home networks has been driven by the emerging A/V (audio/video) technologies like HD (high def- inition) video and QoS (quality of service) technologies (12). It appears to be a large demand in the home network area for the use of applications such as real-time communication with A/V appliances, multimedia album, and high-quality streaming service, all of which requiring QoS. Likewise, as A/V services have become popular in the home networks, there has been much interest in using home net- working technology to provide QoS-guaranteed A/V services. Basically, the demand for QoS-guaranteed A/V services like contents transferring and real-time communication with A/V appliances may exceed the bandwidth or latency constraints of the home network or portions of the home network. Excess demand may cause delivery delay. Since delivery of some of digital media files are time sensitive, delivery delay leads to rendering errors and artifacts that are annoying to users. It has been proposed that reservation or allocation schemes be used to control the demand of network bandwidth (1), (13). Unfortunately, reservation schemes has some limitations. These approaches are based on the premise that the entire system of user, applications, and digital media files may be completely controlled. These approaches require all users and applications to accurately know the amount of network resources they will need priori. It requires that all applications participate in the reservation system so that accurate allocation and calculation can be performed. It, also, can not manage actual network resource status, when applications that are not in reservation system consume network resource. These limitations have discouraged practical implementations. Thus, based on standard home networking protocol, we propose middleware approach that can improve existing reservation-style approaches (especially UPnP QoS architec- ture v1.0) and try to provide the QoS-guaranteed A/V services based on monitoring and adaptation. The proposed approach

In this paper, we introduce a dynamic admission control (DAC) to protect and maintain QoS-demanding traffic streams in IEEE 802.11e enhanced distributed channel access (EDCA) based wireless home network. The proposed DAC dynamically adjusts to achieve high bandwidth utilization while maintaining the desired QoS in a deteriorative channel condition. The ITU-T H.263++ scalable video codec and NS-2-based simu- lation results show that the proposed DAC can efficiently balance the tradeoff between resource utilization and QoS guarantee in unstable wireless home network.

With the advent of high-performance networks, multi-party collaborative environments are widely used to large-scale distributed meetings, collaborative work sessions, and distance lectures. To improve quality of experience in advanced collaborative environments, we attempt to better understanding of human factors by developing high-resolution tiled display to show multiple high-quality video streams simultaneously. It is to show DV (digital video) and HDV (high-definition digital video) on the tiled display consisting of arrays of LCD monitors to form a 6 by 4 display. Consequently this is primarily applied to provide properly large-scale view of collaboration space and detailed close-up of meeting documents. We also propose 3D graphics model sharing service based on hand-like input device. This is to share 3D graphic models in multi-party collaborative environments by using glove-based hand motion tracking and enable to transfer the hand motion to other participants through networks. Thus, this gives convenient interaction method for sharing 3D models.

In the QoS (quality of service) provisioned Internet, resource reservation is required to guarantee the allocation of resources. Traditionally, responding to each reservation call, an admission control is performed to guarantee the exclusive use of resources from the pool of resources without sharing. However, different types of reservations - immediate and advance reservations - are existing and making different requests with varying terms (i.e., session duration may be specified or it may not). When the reservation requests are made in unclear terms, there exists conflict to the network service provider on how to share its resource to both types of reservations. To suggest a balanced solution, in this paper, we propose a revenue-based resource management scheme (focusing on the bandwidth) that effectively regulates resource allocation between immediate and advance reservations. By dynamically adjusting the resource boundary, the proposed resource management scheme adaptively reacts to the changes of underlying networks in a timely manner. Several NS-2 based network simulations show that the proposed scheme achieves more stabilized resource utilization while improving the efficiency of resource sharing.

In this paper, we propose an adaptive event synchronization control for distributed virtual environment (DVE) systems connected over dynamically time-varying networks. Existing adaptive synchronization approaches still have two challenges: disturbance by network traffic load fluctuation and event skipping by contracting the playout delay. Thus, by adopting so-called adaptive event synchronization (AVS) control, we attempt to improve the adaptability of DVE synchronization control. With the proposed control, the event buffer management provides a cost-effective guideline on timing based on idle time for contracting playout delay to suppress the event skipping. Simulation results are presented to show the enhanced performance of the proposed adaptive event synchronization control

A solution for Internet video streaming with two major components is examined in this paper. First, a new rate adaptation mechanism called the smooth and fast rate adaptation mechanism (SFRAM) is developed based on the transmission control protocol (TCP) throughput equation. SFRAM not only achieves TCP-friendliness but also provides a smooth and fast rate adaptation for transmitted video. By adaptively averaging measurements such as the round-trip time and the packet loss rate over a suitable window, SFRAM mitigates fluctuation that is undesirable for video transmission. The adopted weighting scheme enables the response in a fast manner only for distinct network variations so that the overall network utilization is improved while sustaining end-to-end video quality. When integrated with active routing support such as random early detection and explicit congestion notification, SFRAM provides an even better performance. Second, an adaptive network-aware error control is proposed to alleviate error propagation due to packet loss. Based on SFRAM's capturing of distinct network variations, various source-level error control options such as reference picture selection, video redundancy coding and I-MB refresh are dynamically selected. Extensive experiments by using both the ns-2 network simulator and the real Internet environment are performed to evaluate the dynamic behavior of SFRAM.

A new inter-client synchronization framework employing a server-client coordinated adaptive playout and error control toward the multicast streaming of stored media is discussed in this paper. Based on the overall synchronization status as well as the buffer occupancy level, the playout speed of each client is manipulated within a perceptually tolerable range. By coordinating the playout speed of each client, the inter-client synchronization with respect to the target presentation time is smoothly achieved. The cumulative feedback for retransmission assisted by both playout speed manipulation at clients and implicit help of a streaming server is also conducted to increase the reliability. Simulation results demonstrate the performance of the proposed multicast streaming framework.

Packet delay and loss are two essential problems to real-time voice transmission over the Internet. Some effort has been involved in packet-level delay jitter concealment and error control. To minimize the impact of packet delay jitter dropping at the receiver, a packet-based Synchronized OverLap-and-Add (SOLA) algorithm was proposed in [3] to play out packets adaptively. By employing the forward error correction (FEC) for network loss recovery, this paper further extends the applicability of audio time-scale modification to an integrated delay/error concealment Performance improvement of the proposed adaptive playout is provided by comparing with the silenced-based, FEC-integrated adaptive playout. Overhead of the proposed concealment is discussed in terms of delay and complexity. We also investigate the impact of stretching-ratio transition on the perceived quality.

Much effort has been involved in packet-level error control and delay jitter concealment for Internet audio applications. A packet-based time-scale modification scheme for speech signals is developed to provide adaptive delay concealment, at the receiver for an Internet voice session in this work. The adaptive playout algorithm strives to minimize packet droppings for late-arrival packets and premature packets while maintaining the end-to-end delay constraint. By stretching the length of voice segments and incorporating silence intervals, the proposed algorithm is able to accommodate fluctuating delays including delay spikes quickly. It is verified by experiments that the proposed adaptive playout algorithm improves the received speech intelligence under a tightly bounded average playout delay.

We explore a methodology to divide an application's data into several categories and send them through different IP differentiated services vices (DiffServ) grades. The issues are motivated by a futuristic infrastructure wherein the Internet service provider charges different prices to packets going through different service grades. This paper discusses how a subscriber of the DiffServ can assign different grades to real-time video packets on the basis of the packets' relative importance under the constraint of communication expenses. For relative DiffServ, we present an adaptive packet forwarding mechanism to preserve the performance ratio among service grades persistently over different time scales. Through experiments we observe how this packet forwarding mechanism improves the the quality of the application that maps the packets into the different service grades. In the experiment the end-to-end performance is evaluated by the error resilient packet video transmission using ITU-T H.263+ codec over a simulated Diffserv network.

This paper presents a futuristic framework for quality-of-service (QoS) mapping between practically categorized packet video and relative differentiated service (DiffServ or DS) network employing unified priority index and adaptive packet forwarding mechanism under a given pricing model (e.g., DiffServ level differentiated price/packet). Video categorization is based on the relative priority index (RPI), which represents the relative preference per each packet in terms of loss and delay. We propose an adaptive packet forwarding mechanism for a DiffServ network to provide persistent service differentiation. Effective QoS mapping is then performed by mapping video packets onto different DiffServ levels based on RPI. To verify the efficiency of proposed strategy, the end-to-end performance is evaluated through an error resilient packet video transmission using ITU-T H.263+ codec over a simulated DiffServ network. Results show that the proposed QoS mapping mechanism can exploit the relative DiffServ advantage and result in the persistent service differentiation among DiffServ levels and the enhanced end-to-end video quality with the same pricing constraint.

Interactive error control to mitigate error propagation from packet loss is one of the serious challenges to Internet video applications to date. Error control scheme performance is highly dependent on how to coordinate the source error resilience options based upon the observed network status, especially for real-time interactive (i.e. unicast) Internet video. In this paper, a smooth and fast rate adaptation mechanism (SFRAM) (Kim et al., Proc. Packet Video Workshop 2000, May 2000), which does TCP-friendly rate adaptation based on the TCP throughput equation, is used to coordinate the network-aware error control (NAEC) scheme for the best end-to-end video quality. Since the adopted smoothing and weighting scheme of SFRAM enables the capture of only distinct network variations, the various source-level error resilience tools such as reference picture selection (RPS), video redundancy coding (VRC) and I-MB (intra-macroblock) refresh can be dynamically switched. To figure out the best operating mode of interactive error control upon the network status (e.g., round trip time and packet loss rate), performance of each error resilience option is first investigated with a virtual packet loss and feedback scenario. Then, extensive ns-2 network simulation with integrated ITU-T H.263+ encoder is conducted to confirm the efficiency of the proposed NAEC with realistic TCP background traffic and feedback while it behaves in a TCP-friendly way. From simulation, SFRAM-driven NAEC demonstrates enhanced error control over fixed error schemes.

In this research, we investigate quality of service (QoS) interaction provisioning between video applications and components of the DiffServ network. QoS interaction is performed through the mapping of video packets based on the relative quality index (RQI), which represents the relative preference in terms of loss and delay, onto the adaptive packet forwarding mechanism in a DiffServ network. To verify the efficiency of the proposed strategy, the end-to-end performance is evaluated through error resilient packet video transmission over a simulated DiffServ network. Results show that the proposed QoS interaction framework results in an enhanced end-to-end video quality at the same cost constraint.

The quality of video transmitted over time-varying wireless channels relies heavily on the coordinated effort to cope with both channel and source variations dynamically. Given the priority of each source packet and the estimated channel condition, an adaptive protection scheme based on joint source-channel criteria is investigated via proactive forward error correction (FEC). With proactive FEC in Reed Solomon (RS)/rate-compatible punctured convolutional (RCPC) codes, we study a practical algorithm to match the relative priority of source packets and instantaneous channel conditions. The channel condition is estimated to capture the long-term fading effect in terms of the averaged SNR over a preset window. Proactive protection is performed for each packet based on the joint source-channel criteria with special attention to the accuracy, time-scale match, and feedback delay of channel status estimation. The overall gain of the proposed protection mechanism is demonstrated in terms of the end-to-end wireless video performance.

A feedback-based Internet video transmission scheme based on ITU-T H.263+ is presented. The proposed system is capable of continually accommodating its stream size and managing the packet loss recovery in response to network condition changes. It consists of multiple components: TCP-friendly end-to-end congestion control and available bandwidth estimation, encoding frame-rate control and delay-based smoothing at the sender, media-aware packetization and packet loss recovery tied with congestion control, and quality recovery tools such as motion-compensated frame interpolation at the receiver. These components are designed to meet the low computational complexity requirement so that the whole system can operate in real-time. Among these, the video-aware congestion control known as a receiver-based congestion control mechanism, the variable frame-rate H.263+ encoding, and fast motion-compensated frame interpolation components are key features. Through a seamless integration, it is demonstrated that network adaptivity is enhanced enough to mitigate the packet loss and bandwidth fluctuation, resulting in a more smooth video experience at the receiver.

A smart interactive video object generation (SIVOG) system for semantic object extraction from image sequences is presented in this work. SIVOG provides a semi-automatic segmentation system, which uses a semantic layer to interact with the user. It interprets user's input and adaptively determine a processing algorithm. The proposed system is evaluated for several typical MPEG-4 test sequences, in which accurate and consistent results are obtained.

Article Free Access Share on SIVOG: smart interactive video object generation system Authors: Ju Guo Integrated Media Systems Center and Department of Electrical Engineering-Systems, University of Southern California, Los Angeles, CA Integrated Media Systems Center and Department of Electrical Engineering-Systems, University of Southern California, Los Angeles, CAView Profile , Jongwon Kim Integrated Media Systems Center and Department of Electrical Engineering-Systems, University of Southern California, Los Angeles, CA Integrated Media Systems Center and Department of Electrical Engineering-Systems, University of Southern California, Los Angeles, CAView Profile , C.-C. Jay Kuo Integrated Media Systems Center and Department of Electrical Engineering-Systems, University of Southern California, Los Angeles, CA Integrated Media Systems Center and Department of Electrical Engineering-Systems, University of Southern California, Los Angeles, CAView Profile Authors Info & Claims MULTIMEDIA '99: Proceedings of the seventh ACM international conference on Multimedia (Part 2)October 1999 Pages 13–16https://doi.org/10.1145/319878.319882Published:01 October 1999Publication History 4citation248DownloadsMetricsTotal Citations4Total Downloads248Last 12 Months9Last 6 weeks0 Get Citation AlertsNew Citation Alert added!This alert has been successfully added and will be sent to:You will be notified whenever a record that you have chosen has been cited.To manage your alert preferences, click on the button below.Manage my AlertsNew Citation Alert!Please log in to your account Save to BinderSave to BinderCreate a New BinderNameCancelCreateExport CitationPublisher SiteeReaderPDF

Recent interest in supporting packetized video applications over the Internet have been fueled by the availability of low-cost video of tolerable quality. However, efficient transmission of video over the Internet is still a challenging problem due to the best-effort service nature of the Internet Protocol (IP). An efficient receiver-based congestion control mechanism (RCCM), specifically suitable for real-time packet video transmission over the Internet, is investigated in this research. In our design, an end system adjusts the sending rate, depending on the perceived network status, to satisfy the bandwidth and packet loss requirements while exhibiting a TCP-friendly behavior.

A novel rate control algorithm consisting of two major components, i.e. a variable encoding frame rate method and a hybrid DCT/wavelet I-frame coding scheme, is proposed for low bit rate video coding. Most existing rate control algorithms for low bit rate video focus on bit allocation at the macroblock level under a constant frame rate assumption. The proposed rate control algorithm is able to adjust the encoding frame rate at the expense of tolerable time-delay. Furthermore, an R-D optimized hybrid DCT/wavelet scheme is used for effective I-frame coding. The new rate-control algorithm attempts to achieve a good balance between spatial quality and temporal quality to enhance the overall human perceptual quality at low bit rates. It is demonstrated that the rate control algorithm achieves higher coding efficiency at low bit rates with a low additional computational cost. The variable frame rate method and the hybrid I-frame coding scheme are compatible with the bitstream structure of H.263+.

A new way to implement the Internet modem video transmission is presented. This system is capable of continually accommodating its bitstream size in response to changing network conditions. The key idea is to adopt an adaptive least mean squares (LMS) controller to orchestrates an H.263+ encoder rate control at the server end as well as a fast frame interpolation at the client end. It is demonstrated that the user datagram protocol (UDP) packet loss is significantly reduced and a smooth video display can be achieved.

Video object segmentation is an important component for object-based video coding schemes such as MPEG-4. A fast and robust video segmentation technique, which aims at efficient foreground and background separation via effective combination of motion and color segmentation modules is proposed in this work. First, a non-parametric gradient-based iterative color clustering algorithm called the mean shift algorithm is employed to provide a robust initial dominant color regions according to color similarity. With the dominant color information obtained from previous frames as an initial seed for the next frame, we can reduce the amount of computational time by 50%. Next, moving regions are identified by a motion detection method based on the frame intensity difference, which helps to circumvent the motion estimation complexity for the whole frame. Only moving regions are further merged or split according to the region-based affine motion model. Furthermore, sizes, colors, and motion information of homogeneous regions are tracked to increase temporal and spatial consistency of extracted objects. The proposed system is evaluated for several typical MPEG-4 test sequences, and it provides very consistent and accurate object boundaries throughout the entire test sequences.

This paper presents a novel error concealment technique based on the discrete cosine transform (DCT) coefficients recovery and its application to the MPEG-2 bit stream error. Assuming a smoothness constraint on image intensity, an object function which describes the intersample variations at the boundaries of the lost block and the adjacent blocks is defined, and the corrupted DCT coefficients are recovered by solving a linear equation. Our approach can be regarded as a special case of Wang et al.'s (1991). However, we show that the linear equation in the proposed algorithm can be decomposed into four independent subequations, requiring much lower computational load and simpler hardware structure than existing algorithms, while providing adequate performances. To develop a generic error concealment (EC) system, the blocks corrupted by the random bit errors are identified by a multistage error detection algorithm. Thus, the proposed EC system can be applied to more realistic environments, such as concealment of random bit error in MPEG-2 bit stream. Computer simulation results show that the quality of a recovered image is significantly improved even at a bit error rate as high as 10/sup -5/.

A number of segmentation algorithms have been developed, but those algorithms are not effective on volume reconstruction because they are limited to operating only on two-dimensional (2-D) images. In this paper, we propose the volumetric object reconstruction method using the three-dimensional Markov random field (3D-MRF) model-based segmentation. The 3D-MRF model is known to be one of efficient ways to model spatial contextual information. The method is compared with the 2-D region growing scheme under three types of interpolation. The results show that the proposed method is better in the aspect of image quality than other methods.

An image-coding technique, in which the discrete cosine transform (DCT) is combined with a classified vector quantization (CVQ), is presented. A DCT-transformed input block is classified according to the perceptual feature, partitioned into several smaller vectors, and then vector quantized. An efficient edge-oriented classifier employing the DCT coefficients as feature for classification is used to maintain the edge integrity in the reconstructed image. Based on a smaller geometric mean vector variance, a partition scheme in which 2-D DCT coefficients are divided into several smaller size vectors is also investigated. Because the distortion rate function (DRF) used is essential for the bit allocation algorithm to perform well, attempts have been made to modify the asymptotic DRF to estimate the performance of real VQs at low bit rates, and the modification is shown to be in good agreement with experimental results. Simulation results indicate that a good visual quality of the coded image in the range of 0.4 approximately 0.7 b/pixel is obtained.>