The work proposes a method for having vehicular users’ (VUs’) Multicast and Broadcast Service (MBS) in 5G cellular network. Unlike the other works that use both the Channel Quality Indicator (CQI) values and the number of VUs associated with each CQI value to calculate each group’s centroid, the proposed method only uses CQI values without considering the number of VUs associated with each CQI value to calculate centroids to prevent undesirable grouping configurations. We argue that the grouping phase should be only based on CQI values; the number of VUs can be considered in the following resource allocation phase, for which more resources should be allocated to the groups that have more VUs. The performance evaluation results show that the proposed method can have the better performance comparing with the other methods, especially in the high-dispersed CQI distribution situation.

Abstract IEEE 802.11ah, which is a wireless network protocol designed for Internet of Things (IoT), devised the Restrict Access Window (RAW) mechanism to tackle the collision problem. However, some RAW’s slots may be overloaded; some RAW’s slots may be lightly loaded or have no load because no stations (STA) allocated in these slots need to access channel. This work adopted (i) the Registered Backoff Time (RBT) mechanism such that each STA can register the backoff time for its future channel access in access point (AP) and (ii) the Claiming RAW mechanism such that STAs are allowed to notify AP that they have uplinked data to transmit. In this way, AP can (1) know which STAs have uplinked data to transmit and thus (2) re-schedule those STAs from the overloaded slots to the slots that (i) are underloaded or (ii) no STAs need to access channel based on STAs’ RBTs in advance. As a result, the proposed Registration-based Regrouping for Load-Balance Channel Access (RRG-LBCA) method can achieve load balance among slots to increase the IEEE 802.11ah network’s performance. The performance evaluation results shown that the proposed RRG-LBCA method can eliminate the collision situation and increase the aggregate throughput.

Abstract IEEE 802.11ah is a new protocol designed for Internet of Things (IoT). IEEE 802.11ah uses a hierarchical ID assignment schema and designs the corresponding channel access method such that stations can access channel alternatively through the four-level’s hierarchical channel access structure. Nevertheless, collisions still cannot be effectively avoided in a dense IOT networking environment using the legacy IEEE 802.11ah. The work proposed the registration-based situation-aware access extension (RSAE) control scheme to avoid collisions, decrease backoff’s waiting time and improve slot utilization through the following two ways: (i) in contrast to the traditional CSMA/CA protocol that generates the backoff time for the next channel access after the collision happened, registering the backoff time to AP for scheduling the next channel access before the collision happens; (ii) extending the channel accessing’s privilege of those stations that did not complete the data transmission/receiving in the current slot to the available time of the next slot. Comparing with the legacy IEEE 802.11ah, the performance evaluation of the proposed RSAE scheme shown that it still can have a better throughput and a lower collision rate when there is a bigger number of stations in a slot.

Summary This paper proposed a mobile edge computing (MEC)‐based point of interests (POIs) downloading and proximate sharing system for a group of people, who belong to the same mobile social networks in proximity (MSN‐P) and whose handheld devices are connected in a tree‐like structure. Using the proposed system, the root handheld device plays the role of an MEC server and is charge of downloading POIs' contents from the remote cloud server using 4G/5G cellular network and then forwarding POIs' contents to other group members' handheld devices using device‐to‐device (D2D) communication. Since the number of connected handheld devices in each handheld device is limited using D2D communication, this work proposed a control scheme called k‐Connection‐Limited and n‐Hop (kCL‐nH) tree topology's construction scheme to achieve the proximate sharing of POIs' contents. Additionally, since the root handheld device consumes more battery power than others, this work proposed a control scheme that has each group member's handheld device to be the root handheld device alternatively. A credit scheme was thus proposed to denote each handheld device's contribution, which is referred for the selecting of the next root handheld device, to achieve the fairness concern. Then, the control scheme, which is executed in the corresponding MEC server, considering both remaining battery power and credit for the switching of the root handheld device was designed. The proposed method was developed using Wi‐Fi Direct over the iOS system. Experimental results have shown some interested results among power consumption, downloaded data volume and fairness among all handheld devices.

Distributed mobility management (DMM) is designed for flattening the network architecture to resolve the problems, e.g., scalability, non-optimal path, and reliability issues, existed in mobile Internet's mobility management. Although Software Defined Network (SDN) has been applied to DMM to enhance the performance of existed DMM-based schemes, the current mobility management schemes still suffer the high signaling cost and handover latency. In this paper, a Software Defined Network (SDN)-based Backward Fast Handover (SBF-DMM) method that can let MN have the higher chance of being in the predictive mode, which can finish the handover preparation processing before MN disconnecting from the current Mobility Anchor and Access Router's (MAAR's) domain, was proposed. Furthermore, SBF-DMM also can support sub-optimal routing through the help of the SDN Controller. To evaluate and verify the proposed method, an analytical model was devised. The performance analysis has shown that the proposed SBF-DMM has the better handover performance than the other methods in terms of signaling overhead, average handover latency and packets loss.

HTTP video streaming over the wireless mobile network is challenging because the wireless mobile networking environment usually suffers fluctuation in available bandwidth and mobile users usually keep moving and playing streaming video simultaneously. To deal with the issue, a Quality Of Experience (QoE)-oriented adaptive HTTP video streaming control method based on MPEG-DASH over the wireless mobile networking environment was proposed in this paper. In order to get better QoE, the proposed method considers both temporal and geo concerns for estimating the available bandwidth in the future and the proposed adaptive streaming control scheme considers buffer level, video quality of the most recently downloaded segment and the estimated bandwidth to decide the video quality for the next downloaded video segment. The proposed method has been implemented in the Android system for the client side and the Linux system for the server side. The experiments shown that the proposed method can improve initial delay time, suspended time, suspended times, bitrate difference per segment, and average bitrate difference considering suspending.

Summary This work adopts a mobile social network in proximity (MSNP)–centric group touring service concept to share geo–points of interest's (POIs') data among mobile users belonging to the same touring group. In a group‐based touring service, a mobile user can create a group in the MSNP that can consist of some clusters in which some members having the 3G/3.5G/4G cellular network service are selected as cluster leaders to enable their Wi‐Fi hot spot functions as mobile proxies. After that, each group member can join one of the clusters by connecting with a cluster leader's Wi‐Fi hot spot. Thereafter, each cluster leader downloads geo‐POIs' data using 3G/3.5G/4G cellular network and then shares the downloaded POIs' data with its cluster members through its Wi‐Fi hot spot. This work proposed a handoff mechanism such that each mobile user can handoff from its currently connected cluster to the other one that belongs to the same MNSP group. The main handoff concern is that each handheld device only allows a limited number of handheld devices to connect with its Wi‐Fi hot spot at the same time. This work proposed a K‐member‐limited MSNP‐based INter Group handoff scheme called K‐MING to tackle the handoff problem. The performance analysis shows that K‐MING for the group touring scenario has better performance for the sharing of geo‐POIs' data downloading in terms of average power consumption and average downloaded data volume from 3G/3.5G/4G cellular network than that of the individual touring scenario.

Proxy Mobile IPv6 (PMIPv6) is a network-based mobility management protocol. Fast Handover for PMIPv6 (FPMIPv6) offers a handover preparation mechanism for MNs to improve the handover performance. It is possible that several MNs are doing the handover processing from the same previous network to the same new network around the same time. However, the current schemes of FPMIPv6 mainly focus on one MN instead of multiple MNs. It causes the problem of many duplicate control messages during handover. In this paper, a Group-Based Handover Control Scheme for PMIPv6 using the ‘Hitch on’ over heterogeneous wireless mobile networks was proposed. The ‘Hitch on’ concept means that multiple nodes can be grouped together to resolve the aforementioned problem. Multiple MNs, which have similar movement behaviors at the same time, are formed as a group, for which only one set of messages for handover processing is used instead of using n sets of messages. From the performance analysis, even if the number of MNs increases, the amount of handover signaling cost, handover latency and packet lost are reduced significantly and the handover procedure can be simplified using the proposed GB-PMIPv6 scheme.

Abstract The Long Term Evolution‐advanced cellular network is designed for human‐to‐human communication. When a large number of machine‐to‐machine (M2M) devices are trying to access the network simultaneously, it leads to a low random access (RA) successful rate and high congestion problem, which may cause the waste of radio resources, packet loss, latency, extra power consumption, and the worst, M2M service error. There is an urge to propose an efficient method for M2M communication on the LTE‐A network to resolve the congestion problem. In this paper, we propose a congestion reduction mechanism, which can analyze and model the RA procedure on the Long Term Evolution‐advanced network, to find out the collapse point in the RA procedure and then design a scheme named device‐to‐device cooperative relay scheme to relieve the congestion problem. Meanwhile, this work also adds a relay access barring algorithm to improve performance and an RA resource separation mechanism for human‐to‐human communication. The proposed method can effectively reduce the network congestion problem. Simulation results show that the network throughput and the congestion can be significantly improved using the proposed mechanism. Copyright © 2016 John Wiley & Sons, Ltd.

Datagram congestion control protocol (DCCP) is an unreliable transport layer protocol with a congestion control mechanism used for multimedia streaming. In order to use multiple network interfaces to transmit streaming data smoothly, a multipath datagram congestion control protocol (MP-DCCP) is proposed and presented in this paper. Video streaming transmission through multipath faces three problems: (1) unordered and untimely received packets at the receiver side, (2) the importance of frames/packets are different and (3) the stability of the total throughput of a multipath transmission. This work proposes the adaptive ordering predicting scheduling (AOPS) scheme to relieve the first two problems. AOPS estimates packets’ arrival orders at the receiver side through these multiple paths and then derives packets’ transmission schedule at the sender side. A flow-sharing congestion control (FSCC) mechanism for MP-DCCP is proposed in this work to tackle the third problem i.e., the total throughput's stability. From the simulation results, we illustrated that the packets’ disorder and received situations in MP-DCCP can be improved using our proposed AOPS scheme and the unstable condition of the total throughput in MP-DCCP can also be relieved using the proposed FSCC mechanism.

With the rapid development of smartphones and personal tablet computers, it brings a greatly growing rate of ubiquitous applications for location-based services (LBS). One famous LBS is the mobile advertisement. A mobile advertisement system brings benefits and opportunities among users, service providers, and advertisers. In this paper, we propose a personalized context-aware mobile advertisement system (PCA-MAS) over cellular networks, which contains two new techniques called (i) approaching detection method (ADM) and (ii) context-aware ad targeting method (CAADTM). ADM can find some point of interests that a user is approaching; CAADTM pushes advertisements that satisfy user's requirement based on the user's context, that is, user's profile, current time, current position, and so on. Our experimental results show that (i) ADM has the good hit rate to determine those point of interests that a user is approaching within the 150-m radius of the approaching range, and (ii) CAADTM has the good hit rate of finding appropriate advertisements that a user prefers through the favorite content table filtering, the annoying content table filtering, and the advertisement clicking feedback. Copyright © 2013 John Wiley & Sons, Ltd.

Abstract To have reliable transmission in the vehicular ad hoc network (VANET) environment, a split reliable transport protocol (SRTP) is proposed in this paper. The basic idea of SRTP is to separate a long connection into several subconnections in order to reduce the overhead of packet retransmission, in which the length of each subconnection depends on links’ states between vehicles. Three main issues that need to be resolved for SRTP are the following: (i) how to decide the length of each subconnection when a connection is established initially; (ii) how to enhance the connection's stability after the connection is established to deal with the fast topology change in the VANET environment; and (iii) how to deal with packet congestion. An estimative scheme using location and speed information of vehicles to estimate the connectivity time of the links between vehicles are used in SRTP. The estimated values can be used to decide the lengths of subconnections and adjust them dynamically. Another mechanism named channel busy rate aware congestion avoidance is proposed to avoid network congestion in the ad hoc network environment. The simulation results show that SRTP can be operated well in the VANET environment, and the aforementioned three issues can be appropriately resolved. Copyright © 2014 John Wiley & Sons, Ltd.

This paper presents a 3.2 Gb/s low-power clock and data recovery (CDR). The improved architecture using two half-rate gated voltage-controlled oscillators (GVCOs) shared between frequency presetting and data recovery modes is presented to remove the LC-tank voltage-controlled oscillator in a cascaded CDR. Moreover, using the proposed active inductive loading technique instead of the on-chip inductor reduces the power consumption and area in high-speed operation. This CDR circuit has been designed in TSMC 0.18 μm CMOS technology. It consumes 22.5 mW from a 1.8-V supply and occupies an active area of 0.26 mm2. The peak-to-peak and rms jitter of the recovered clock are 95.6 ps and 12.1 ps for a 3.2 Gb/s 27-1 PRBS, respectively.

Video streaming over the wireless mobile network suffers from signal fading and shadowing effects. For instance, when the train is moving in a fixed route over the railway, the signal will vary along the moving route, especially on some specific positions, e.g., tunnels and underground. This serious signal variation results in packets loss and affects the video presentation quality. Furthermore, if the length of the tunnel is too long, the streaming buffer at the client side will have the starvation phenomenon. In this paper, we proposed a video streaming control mechanism called Location-Based Video Streaming Control mechanism for Trains (LBVS-T). Since the characteristics of the train are moving on a fixed route, the signal variation will show a specific pattern. The proposed mechanism can adjust the transmission rate based on the network disruption situation, which is caused by the geographic environment. The performance results demonstrate that LBVS-T can tackle the buffer starvation problem and have more smooth streaming performance when a train is moving before entering into, inside and just outside tunnels.

When a number of persons, e.g., a family or a group of friends, drive their vehicles for a trip together, they can form a fleet of vehicles and share their network resources during their trip. Let one member want to watch a video from the Internet. He may not have high resolution or video quality due to his limited 3G/3.5G bandwidth to the Internet. The cooperative video streaming scenario allows the requested member to ask other members of the same fleet to download video cooperatively. In other words, other members can help to download parts of the video from the Internet and then forward video data to the requested member hop by hop through the ad-hoc network. This work proposes the k-hop cooperative video streaming protocol using H.264/SVC over the hybrid vehicular networks which consist of 3G/3.5G cellular network and Dedicated Short-Range Communications (DSRC) ad-hoc network. In order to smooth video playback over the DSRC-based ad-hoc network, this work proposes: (1) one streaming task assignment scheme that schedules the streaming task to each member over the dynamic vehicular networks, and (2) packet forwarding strategies that decide the forwarding sequence of the buffered video data to the requested member hop by hop. Finally, we utilize the network simulator version 2 (NS2) to simulate the proposed protocol. Based on the simulation results, the proposed scheme can estimate the assignment interval adaptively and the playback priority first (PPF) strategy has the best performance for the k-hop video forwarding over the hybrid vehicular networks.

ABSTRACT Network mobility (NEMO) enables a group of nodes to connect to the back‐end infrastructure during the movement. Because a vehicle must provide passengers good enough transmission quality, various access techniques, for example, 3 G/3.5 G, Wi‐Fi or dedicated short range communication, can be simultaneously equipped to aggregate bandwidth and tolerate link failure. This paper extends the Session Initiation Protocol (SIP)‐based NEMO framework to support multihoming. First, IEEE 802.21 Media Independent Handover (MIH) is integrated to manage multiple egress interfaces in an on‐board unit (OBU). Second, a dynamic synchronization mechanism using the SIP SUBSCRIBE and NOTIFY methods is proposed to manage multiple OBUs in a vehicle. Furthermore, the framework is applied to public transportation systems, for example, bus, train, or mass rapid transit. Passengers may need to transit several times during their journeys. The transit behavior must induce a flood of signaling messages for location update. In this paper, the SIP REFER method is exploited to reduce signaling messages while a group of passengers have scheduled their transit in advance. Copyright © 2011 John Wiley & Sons, Ltd.

Although many kinds of collision pre-warning algorithms and systems for vehicles were proposed in order to prevent accidents in past years, two problems have to be resolve in order to enhance the accuracy of collision detection. First, a vehicle does not always keep the same velocity when it is moving. Second, the curve road condition should be considered. In this paper, we proposed an Advanced Cooperative Collision Warning algorithm named ACCW to deal with these two problems. Basically, ACCW uses the acceleration parameter to correct some errors that are caused by velocity and direction changing. The simulation results show that that the proposed ACCW algorithm has high accuracy's warning rates in both intersection and curve situations.

With the development of vehicular networks, new types of networking services arise for automotive users. Since vehicles have high mobility, it is difficult to keep stable and reliable network connections between vehicles. With regard to multi-media streaming services, a vehicle may not have good QoS of video streaming using a single 3G/3.5G wireless interface. This paper is motivated to investigate the cooperative streaming sce-nario in a fleet-based vehicular network. When a vehicle requests a video stream encoded in H.264/SVC from the Internet using its 3G/3.5G network, it asks for nearby vehicles belonging to the same fleet to download the requested video data cooperatively through these nearby vehicles' 3G/3.5G network and then forward these downloaded video to the requested vehicle through V2V Dedicated Short-Range Communication (DSRC) networks. Since the topology of a vehicular network may change dynamically, an adaptive video transmission scheme is utilized to control the sending rate of the server to avoid the congestion of vehicular networks and retransmit the lost video to the requester to improve the quality of video before the playback deadline. We utilize EstiNet Network Simulator and Emulator (NCTUns) to simulate the proposed schemes. Based on the simulation results, the proposed priority first retransmission scheduling (PFRS) can have higher PSNR values in the network condition with lower Bit Error Rates (BER) and the dependency cost retransmission scheduling (DCRS) would have higher PSNR values in the network condition with higher BER.

In this paper, we proposed a three-tier platform for providing group streaming service over mobile network environments with Fixed Mobile Convergence (FMC) techniques. The upper layer server takes responsibility of Quality of Service (QoS) computing and scheduling for serving heterogeneously grouped streaming requests. The middle layer servers take care of streaming packets' collection, packetization, and transmission in real time. The low layer servers are allocated on public transportations for dealing with the multiplexed streaming packets' depacketization and distribution. In order to efficiently utilize bandwidth during group streaming services, we developed an efficient MSE-based multiplexing mechanism for analyzing and scheduling multiplexed streaming packets' transmissions. Comparing with traditional method of establishing direct end-to-end connection for each streaming request, the performance results of the proposed platform demonstrate that the developed multiplexing mechanism can reduce almost 40% variation of required bandwidth, and it can successfully serve grouped streaming requests in an efficient way.

SUMMARY In this paper, a distributed telematics peer‐to‐peer networking system is proposed to provide an efficient and feasible service discovery mechanism for mobile users. When mobile users travel on the road, they may request for some services from service providers to meet their demands. Mobile users in vehicles are assumed to go through a lot of regions, in which each region is associated with a region server. Related information of all service providers in a region is stored in its region server, and each region server is in charge of the service discovery requests from all vehicles located in its region. This paper focuses on the issues of developing a distributed peer‐to‐peer telematics service discovery networking system over the vehicular network environment for mobile users. To provide the communication mechanism for telematics service discovery among region server, mobile users, and service providers, the Telematics Service Markup Language is proposed. According to clients' demands, three types of query scenarios are proposed, which are (1) reference point based query, (2) continual query, and (3) route‐based query. Finally, we present usage examples and system implementation of distributed telematics peer‐to‐peer. Copyright © 2012 John Wiley & Sons, Ltd.

In this paper, we proposed a Proactive Member-Centric routing protocol using the Several strategy (PMCS) for having bandwidth aggregation in the hybrid VANET environment such that a better collaborate video streaming service in the "fleet scenario" can be achieved. We consider a group of vehicles that drive from the same starting point with the same route and are going to the same destination. The main idea of this paper is to make use of these vehicles' driving patterns and several strategy to provide a more efficient bandwidth aggregation way for streaming applications in VANET. PMCS maintains the topology among fleet member nodes proactively. By giving member nodes higher priority to forward the data, the special driving pattern exists in fleet members is going to benefit the routing because the links between member nodes are more stable than others, by maintaining member nodes and suitable nonmember nodes, PMCS can reduce some control overhead, by inducing a scheme to find suitable nonmember nodes to help forwarding, PMCS can connect the separated topology. To evaluate the performance of the proposed protocol, we also address a fleet mobility model. The simulation results show that our proposed routing protocol has higher throughput and packet delivery ratio with much lower control overhead.

When a passenger in a vehicle wants to watch video during movement, he may request video through cellular network, e.g., 3G or 3.5G network. However, due to the characteristics of vehicular networks, the limited bandwidth of 3G/3.5G network is not enough to support high resolution or high video quality. In order to improve the quality of video playback, this vehicle, which is defined as requester in this paper, may try to ask other members of the same fleet to download video cooperatively. In other words, other members would download parts of requested video through their 3G/3.5G network individually and then forward video data hop by hop to the requester using the DSRC ad-hoc network. In order to smooth the DSRC ad-hoc network, we propose and discuss two k-hop packet forwarding schemes for the aforementioned cooperative video streaming over vehicular networks in this paper. Each intermediary node of the forwarding route must decide how to deliver buffered video data through the limited DSRC network. Finally, the NS2 simulation tool is adopted to evaluate the proposed schemes. Based on the simulation results, the Playback Priority First (PPF) scheme is performed better than the First-In First-Out (FIFO) scheme.

When a vehicle requests a video streaming service from the Internet, it may not have enough bandwidth to receive good video quality. In order to resolve this problem, the vehicle is proposed to ask neighboring vehicles to help the video download process using a short-range wireless technique, i.e. Dedicated Short Range Communications (DSRCs). In this paper, this scenario is called cooperative streaming over vehicular networks. Since the topology of vehicular networks changes over time rapidly, three critical issues in the moving-vehicle-based cooperative streaming are: (i) how to form a cooperative group dynamically, (ii) how to schedule neighboring vehicles to help the video download process and (iii) how to estimate the possible change of network conditions. We propose corresponding control schemes to tackle the aforementioned issues in this paper. From the simulation results, we can find that different scheduling schemes are adaptive to different scenarios, and the proposed dynamic group formation scheme has better performance than that of periodic group formation scheme.

The link performance is highly unpredictable and the connections are broken frequently in opportunistic networks (Oppnets). The transmission control protocol will perform badly in Oppnets because the end-to-end connection always needs to be reconstructed and data need to be retransmitted from the original sender again through the reconstructed connection. To solve this problem, the study investigates the principles of on-time and at-one-time delivery, in which ‘on-time’ means to transmit data to the receiver during a pre-scheduled time period and ‘at-one-time’ means to transmit each piece of data from the sender just once. Based on the principles, the designed adaptive transmitting platform (ATPOO) supports the on-time and at-one-time delivering capacities in Oppnets. In ATPOO, the authors separate the transmitting process into macro- and micro-stages. The macro-stage deals with the end-to-end transmission quality of service; the micro-stage focuses on the transmission between adjacent nodes along with the pre-constructed delivering path. The proposed ATPOO based on the delay-tolerant network (DTN) has been implemented in this study. The experimental results show that the designed ATPOO performs the highest successful delivery ratio of 97% over complex and unpredictable network environments while the ratios of best effort, S-FTP and DTN methods are 81, 76 and 89%, respectively.

SUMMARY The rapid progress of information techniques enables people to have a relatively convenient environment and life. This study designs and develops the Web2.0‐based Home Appliances controL sErvice platform (WHALE) for users to easily control home appliances. WHALE is a three‐tier architecture, which consists of Device Detection and Registration Service Platform (DDRSP), Web2.0 Service Platform (WSP), and Common Service Platform (CSP). WHALE is based on the standards of Devices Profile for Web Services (DPWS), Open Services Gateway initiative (OSGi), and Web2.0 techniques. Users can use many kinds of handheld devices to get services from WSP. When DDRSP finds a new home appliance, this home appliance will be registered into CSP. CSP records the related information and services which are provided by this home appliance. WSP not only integrates and shows the related services of all home appliances but also provides controlling interfaces to control the actions of all home appliances. All delivered messages are based on Simple Object Access Protocol (SOAP) and Web Services Description Language (WSDL). Furthermore, for the Web QoS (Quality of Service) strategy, this study designs and proposes an Intelligent Web qualIty of Service‐aware maTching (iWIST) model to satisfy different characteristics of the user's requirements. iWIST consists of Components Classifier, Admission Controller, and Dynamic Content Adaptor. Component Classifier can classify users' requesting components with different QoS constraints. Admission Controller determines the maximum available components for the user. Dynamic Content Adaptor dynamically delivers component information to users. Experiment results illustrate that WHALE can provide appropriate contents with low response time to users. Finally, WHALE has integrity, scalability, operability, adaptivity, and extensibility functions. It can be a reference model for developing home appliance controllers in the future. Copyright © 2011 John Wiley & Sons, Ltd.

In this paper, we proposed a K-hop bandwidth aggregation scheme for member-based cooperative video streaming over the hybrid vehicular networks consisting of DSRC ad hoc network and 3G/3.5G cellular network. Each member in the same group is known to each other, e.g., they may be a family or a group of friends driving their sedans and they set out from the same position along the same path to the same destination during the journey. When a member in a vehicle wants to watch video during traveling, he will request video through 3G/3.5G network. However, the limited bandwidth of the 3G/3.5G network is not enough to support higher resolution or higher quality of downloading video. In order to enhance the quality of video playback, the member, who is called requester, in one vehicle can ask members, who are called helpers, of the same group in other vehicles to cooperatively download video through their 3G/3.5G networks and then forward their downloaded video to the requester through the DSRC ad hoc network. In order to select suitable members as helpers to download and forward the video and make video streaming more effectively to the requester over the DSRC Ad hoc vehicular network, how to select suitable helpers to forward the downloaded video hop-by-hop back to the requester needs to be resolved. In this paper, we proposed a K-hop bandwidth aggregation scheme to tackle the aforementioned problem. The performance results show that our proposed scheme can get better quality video than cooperative video streaming schemes without the deliberately designed K-hop bandwidth aggregation mechanism.

Due to the advance of wireless and mobile technologies, vehicular networks are widely discussed in recent years. When a vehicle requests a video stream from the Internet using the cellular network, i.e., 3G or 3.5G network, it may not have enough bandwidth to receive good quality of video. In order to resolve this problem, the vehicle can ask neighboring vehicles for the help of downloading the video stream. This scenario is called cooperative streaming in vehicular communications. The vehicle can communicate with neighboring vehicles using a short-range wireless technique, e.g., Dedicated Short Range Communications (DSRC). Since the topology of vehicle networks changes rapidly, two critical issues in the moving vehicle-based cooperative streaming are (1) how to form a cooperative group dynamically and (2) how to schedule neighboring vehicles to help the video downloading process. The simulation results reveal that the improvement of video quality is proportional to (1) high traffic flow, (2) high probability of willingness to share bandwidth and (3) low relative speed.

The authors propose a partially reliable-concurrent multipath transfer (PR-CMT) protocol for multimedia streaming. The novelty of PR-CMT is to combine techniques of CMT's concurrent multipath transfer, PR-SCTP's partially reliable transmission and prioritised stream transmission. Moving picture experts group (MPEG) frames are with varying priorities and lifetime. PR-CMT can transmit important frames before other frames but does not transmit lifetime expired frames. The combination of the aforementioned techniques, however, causes imprecise congestion window and receiver buffer blocking problems without suitable control. Consequently, the throughput and video quality degrade. PR-CMT adopts the concept of delay abandoning data to resolve the imprecise cwnd and the receiver buffer blocking problems. The simulation results show that PR-CMT can precisely infer cwnd. Most importantly, PR-CMT can prevent large gaps between two playable frames to have good video quality.

In this paper, we propose a new cache management scheme in the proxy server for location based services. The proxy server is able to cache query results of location-based services (LBS). One of the major LBS is mobile advertising. Mobile advertising is to distribute advertising messages to the target users with mobile devices. Nowadays, users are willing to receive mobile advertising messages. Users can issue queries to search advertisements within specific area. The searching area of the user query is seen as a rectangle in this paper. In traditional cache management, even though the searing areas of different users' queries are similar geographically, users' queries are still viewed as two different queries. The cache performance is low. In this paper, we propose the concept of the geographically partial matching for LBS. In geographically partial matching, two different queries are viewed as the same query if their searching areas are similar. As a result, the hit rate in the cache of the proxy server for LBS can be improved. The simulation results show that the partial matching can increase the re-utilization ratio of the cached content dramatically.

With the maturation of wireless networks technologies, Telematics service is becoming a new rising and popular network service and application. In this paper, a centralized vehicular Peer-to-Peer Telematics (C-VP2P) service discovery platform is proposed to provide efficient and feasible service discovery. This paper focuses on the issues of developing a Telematics service platform over the vehicular network environment. When a user needs a service on the road, two issues need to be considered: (1) how to have the candidate list of service providers? (2) how to select suitable service providers from the candidate list? In order to tackle these two issues, we propose a C-VP2P-based system service discovery mechanism using the vehicular P2P messaging protocol. In this paper, the vehicular P2P messaging protocol is designed to communicate among Telematics central server, clients and service providers. In addition, a service discovery mechanism in the C-VP2P system is proposed to achieve high precision discovery for service providers. We also exhibit the performance merits of the proposed C-VP2P Telematics service discovery platform in this paper.

A Cooperative Collision Early-Warning System (CCEW) was proposed in this paper to tackle the Driving Safety concern. CCEW is a pre-warning algorithm that can be used in the curve environment. In CCEW, vehicles broadcast Relationship Information (RI) to neighboring vehicles through wireless communication (e.g. DSRC) periodically. Each vehicle calculates whether potential collision may happen or not upon the RI messages that were received. After calculation, the CCES system sends a warning message to the driver if the collision between two vehicles may occur. The results show that the warning accuracy rate of CCEW outperforms others in the curve environment.

Typically, a vehicle uses On Bus Unit (OBU) to connect to Road Side Unit (RSU) in vehicular networks. When a Corresponding Node (CN) communicates with a MN (Mobile Node, i.e., a vehicle) in vehicular networks, packets are transmitted through both the wired and wireless links. The link between OBU and RSU is a wireless link, while the link between RSU and CN is a wired link. The wireless link is known with a high packet loss rate. Many packet losses in the wireless link are caused by error loss. If the retransmission is made by CN, CN needs to retransmit packets through both the wired and wireless links. Thus, in this paper, we proposed to use the relay gateway approach for having SCTP (RG-SCTP) in vehicular networks. The relay gateway is able to retransmit packets caused by error loss in the wireless link. The retransmission need not be made by CN. As a result, retransmission made by the relay gateway is much faster than that made by CN. The simulation results show that RG-SCTP can have faster transmission speed and less wasted bandwidth in the wired link than the basic SCTP can.

Internet protocol multimedia subsystem (IMS) is an overlaid infrastructure above the packet-switched domain of the universal mobile telecommunication system core network. IMS enables and enhances real-time multimedia mobile services. IMS signaling and media are separated in IMS. For media session negotiation, two basic procedures in IMS are (i) IMS registration procedure and (ii) IMS incoming call setup procedure. IMS signaling needs to traverse a set of call session control functions (CSCFs), which are proxy-CSCF, interrogating-CSCF (I-CSCF) and serving-CSCF (S-CSCF). However, we observe that I-CSCFs need to query the home subscriber server (HSS) to obtain the name of S-CSCF for each subscriber's request in both the basic procedures. It results in a certain load occuring in the HSS and a certain procedure delay also occurs. Therefore, we employ the distributed hash table (DHT) into IMS to form a DHT-based I-CSCF network. Each I-CSCF in the DHT-based I-CSCF network can share and relieve HSS's load. As a result, the incoming call setup procedure can be speeded up. We also propose two maintaining procedures to maintain the DHT-based I-CSCF network for I-CSCFs join and leave scenarios. We propose one maintaining mechanism to guarantee the proposed system to be able to work well in the failure scenario of I-CSCF. According to the performance evaluation results, the load of HSS is reduced and shared among I-CSCFs in the home network and the signaling delay cost of the incoming call setup procedure is less than that of existent systems even if failure occurs and then is removed.

As more and more devices are equipped with multiple network interfaces, the multihomed technology is becoming an important technology that can provide more reliable connection against the network failure and higher data transmission rate. Concurrent Multipath Transfer (CMT), which is based on SCTP, is devised for multihomed networks. CMT can transmit data over all paths to maximize the data transmission speed. One of the main concerns of CMT is the receiver buffer blocking issue, especially in multihomed wireless mobile networks. Receiver buffer blocking is mainly caused by lost data. In this paper, we propose mobile CMT (mCMT) over wireless networks to deal with the receiver buffer blocking caused by error loss and handoff loss. mCMT has the features of (i) path-oriented multistreaming, (ii) MIH-assisted handoff mechanism, in which MIH denotes Media Independent Handover and (iii) association resume. Path-oriented multistreaming can reduce the receiver buffer blocking caused by error loss. MIH-assisted a handoff can reduce the amount of packet loss during a handoff, which may lead to receiver buffer blocking. Moreover, mCMT introduces the concept of association resume to recover the disconnected association. Simulation results show that mCMT can reduce the effect of the receiver buffer blocking effectively for devices that have smaller buffer size.

This study presents a Web2.0-based Omnibearing Homecare System (Web2OHS) that uses our proposed Application Layer Somecast (ALS) protocol for real-time interactions. Web2OHS provides omnibearing homecare and patientcare services for medical staff and caregivers, which are capable of assisting families, physicians, and nurses to obtain patients' physiological information using healthcare sensors, as well as monitor their behaviors using monitoring-based services. Web2OHS is a three-tier architecture that consists of Web2OHS clients, the Web2.0-based Patientcare Service Platform (WPSP), and the Medicine-based Active Database (MADB). Users can interact with Web2OHS using various appliances and retrieve the latest physiological and monitoring data using really simple syndication (RSS) and the proposed ALS services. The WPSP supports Web2.0-based applications, including blog-like monitoring services, monitoring-based RSS services, a real-time interaction services, and ALS services. The MADB provides a well-designed database, which stores physiological information, clinical information, digital imaging and communications in medicine image files, and monitoring frames. All of the delivered messages are based on extensible markup language and the Health Level 7 protocol. The proposed Web2OHS can support medical informatics and is compatible with related medical information systems.

This study designs and proposes a blog-based interactive patientcare system, called BiPS. BiPS provides omni-bearing patient services. It assists families, doctors and nurses to care patient's physiological information using healthcare sensors and to monitor patient's behaviors using the monitoring-based services. BiPS is a three-tier architecture that is composed of BiPS clients, the Web2.0-based service platform (WSP), and the HL7-based medical database (HMD). Users can interact with BiPS using heterogeneous appliances and get related latest physiological and monitoring information using our proposed application layer somecast (ALS) service. The WSP provides Web2.0-based applications, including blog-based templates and interfaces, monitoring-based RSS services, interaction services, and ALS services. The HMD provides well-designed triggers and a database, including clinical information, physiological information, DICOM image files and monitoring frames. All of the delivered messages are XML-based using the HL7 protocol. The proposed BiPS can support medical information services and can be compatible with related medical information systems.

In this paper, we propose a combined SVC-MDC (scalable video coding & multiple description coding) video coding scheme using the multi-core parallel programming paradigm for P2P video streaming, which is denoted Co-SVC-MDC. To date, P2P video streaming applications are widely popular and emphasized, e.g., PPstream and PPlive, because of higher transmission speed and data availability. However, in the heterogeneous P2P network environment, users are able to utilize PDA, notebook or desktop computer through distinct network interfaces to get on-demand videos ubiquitously. To provide distinct spatial-resolution/fidelity videos and flexible video transmission (playback) over P2P networks, a brand new coding architecture needs to be devised. In the proposed Co-SVC-MDC coding scheme, distinct MDC descriptions contain distinct portions of raw video frames, and each raw frame can be compressed as base layer and SVC enhancement layers. In our experiments, a real implementation of Co-SVC-MDC is exhibited and corresponding performances, e.g., PSNR and decoding speeds, are compared with original SVC in distinct congestion-level P2P networks.

When a group of users or devices move together, network mobility resolves how to manage the location transition of the whole members. Two kinds of solutions have been proposed at different layers. Network mobility (NEMO) basic support protocol (BSP) is the extension of mobile IPv6 at the network layer. On the other hand, Session Initiation Protocol (SIP) based network mobility (SIP-NEMO) is defined at the application layer. However, on the perspective of ubiquitous network environment, a mobile router requires to possess several egress paths to the Internet for providing uninterrupted services and aggregating good enough bandwidth. Multihoming in network mobility support becomes one urgent and necessary issue. Since Mobile IPv6 based NEMO BSP, called MIPv6-NEMO hereafter for convenience, uses a bi-directional tunnel between home agent and mobile router, Multihomed MIPv6-NEMO would suffer and deepen header overhead and suboptimal routing. In the past, we have proposed SIP-NEMO to achieve route optimization. In this paper, we investigate and analyze how to extend the current SIP-NEMO to support multihoming further. The proposed multihomed SIP-NEMO is simulated in different cases and integrated with the IEEE 802.21 Media Independent Handover (MIH) standard. According to the simulation result, the SIP-NEMO can keep route optimization no matter which multihomed case is configured.

Network mobility is proposed to let a mobile network change its point of attachment and still keep all nodes attached to the mobile network globally reachable. However, due to finite bandwidth, limited signaling coverage and frequent link failure, a mobile network needs multihoming to access the Internet simultaneously on the perspective of performance and reliability. In this paper, we extends our previous work, i.e., SIP-based network mobility (SIP-NEMO), to support multihoming. A mobile network is considered to possess more than one egress gateway. The proposed multihomed SIP-NEMO provides dynamic synchronization among different egress gateways. Once an egress gateway joins or leaves the mobile network, the multihomed SIP-NEMO can establish or switch sessions on another alternative egress gateway. According to the simulation results, the proposed multihomed SIP-NEMO can achieve route optimization for data transmission.

As more and more devices are equipped with multiple network interfaces, the multihomed technology is becoming an important technology that can provide more reliable connections against the network failure and better data transmission rate. Concurrent Multipath Transfer (CMT), which is based on SCTP, is devised for multihomed networks. CMT can transmit data over all paths to maximum the data transmission speed. One of the main concerns of CMT is the receiver buffer blocking issue, especially in multihomed wireless mobile networks. If the receiver buffer is blocked, the sender can send only one packet to the receiver. The receiver buffer blocking is mainly caused by the lost data. For small memory devices, e.g., PDA, smart phone and embedded systems, the receiver buffer is limited and has less free space to prevent receiver buffer blocking. In this paper, we proposed Media Independent Handover - Concurrent Multipath Transfer (MIH-CMT) over wireless networks to deal with the receiver buffer blocking caused by handoff loss. MIH-CMT can reduce the amount of packet loss during handoff, which may lead to the receiver buffer blocking. Simulation results show that MIH-CMT can reduce the effect of the receiver buffer blocking effectively for devices that have smaller buffer size.

In this paper, we present ReWarn, an opportunistic relay scheme for cooperative collision warning in VANET, which helps to mitigate the impact of package loss caused by shadow fading. When safety messages in VANET, e.g., collision warning messages, are broadcast in air, a vehicle may detect collision risks both of its own and of its neighbors'. If a vehicle has detected its neighboring vehicles are experiencing a collision risk while they keep quiet, it then probabilistically relays the raw data received from the neighboring vehicles with its own detection. Simulation results show that ReWarn can achieve higher reliability in message transmission than the non-relay schemes and keep the transmission efficiency at a satisfied level as well.

In order to adapt the quality of an on-demand video stream over a time-varying bandwidth channel, a network-aware bandwidth estimation and rate control scheme are required. This paper proposes a predictive video-on-demand (VoD) bandwidth management and a feedback-based buffer control scheme for streaming fine granular scalability videos over wired/WLAN/3G networks. The predictive VoD bandwidth management includes two parts: bandwidth estimation and rate adaptation. According to the measured information of packet round-trip-time, loss-rate, delay jitter and received bit-rate, an improved Kalman filter is proposed to predict an available bandwidth recursively, and to determine a proper transmission rate in consideration of buffer fullness of a decoder. The optimal parameters of the Kalman filter, e.g. a transition matrix and error covariances, can be initialized, converged and adapted to characteristics of the current network. In our experiments, distinct network traffic models are simulated in comparison with pathChirp and one Republic of China patent. The corresponding estimation results with respect to network information are also exhibited in the real networks.

Energy conservation and network performance are critical issues in wireless sensor networks. The authors present a novel efficient communication topology control protocol, called quorum-based load-sharing control protocol (QLSCP). QLSCP is a quorum-based communication protocol, which chooses appropriate communication nodes, adjusts the service loads of critical nodes and performs adaptive sleep management. QLSCP is suitable for harsh environments without a central control server calculating the locations of sensors, using the factor of the remaining power to build the system topology. The proposed protocol divides the topology operation into topology formation, adjustment and execution phases. The topology formation phase builds the backbone of an enhanced tree. In the topology adjustment phase, the enhanced tree is adjusted by an optimal balance of critical nodes in the backbone. In the topology execution phase, the efficient surplus-energy consuming (ESC) mechanism is proposed to efficiently exhaust the energy of each node. The ESC mechanism is designed to efficiently exhaust the latest remaining electronic power of sensor nodes. Simulation results of QLSCP demonstrate that it efficiently achieves to prolong system lifetime in harsh environments.

Synchronous audiovisual streaming and playout are two of the major issues in the multimedia communication network. However, the past corresponding researches of media synchronization mainly focused on the mono-quality and single-layer (nonscalable) audiovisual data. To overcome challenges of ubiquitous multimedia streaming, a scalable audiovisual coder that can provide flexible scalabilities and adaptive streaming control to adapt to complicated network situations are both required. This paper proposes a multilayered audiovisual streaming scheme to deliver layered audiovisual data synchronously, which is called ML-AVSS. Fine-granular scalability (FGS) and bit-sliced arithmetic coding (BSAC) techniques are used to segment video and audio data into one base-layer and multiple enhancement-layer bitstreams. With advantages of audiovisual layer coding, a de-jitter procedure, a conditional retransmission mechanism and a playout synchronization mechanism are designed to transmit hybrid multilayered audiovisual bitstreams in consideration of the result of a network bandwidth adaptation and the distinct decoding time-complexity. Experimental results show that the proposed ML-AVSS is a feasible streaming scheme to overcome challenges of ubiquitous multimedia streaming, e.g., constrained channel bandwidth, quality degradation, unsmooth playout, etc.

Abstract Mobile IP (MIP) defines a mobility management for mobile users to continuously access data when the currently attachment is changed to another network. However, when mobile node (MN) roams between network segments, the handoff latency results in packet losses and transmission delay. In this paper, we propose a multihomed fast handoff scheme (MFH‐MIP) to decrease the handoff cost. In the proposed MFH‐MIP scheme, each MN is implemeted with the link layer trigger and multihomed techniques. Based on the link layer trigger, MN can collect signal strengths of nearby access points (APs) and switch to a new link automatically when the old link becomes unsuitable to connect. Using the multihomed technique, MN can prepare for handoff using two (or more) interfaces, in which (i) one is connected with the old link to receive packets and (ii) the other one is used to access nearby APs and select the most suitable one as the new link, in parellel. In this way, MN can continuously transmit and receive packets during handoff. Based on the proposed method, MN can roam smoothly among different networks in the wireless environment. Copyright © 2008 John Wiley & Sons, Ltd.

Abstract The emergence of centralized WLAN architectures is proposed to solve the problems of large scale deployment of WLANs. In the centralized WLAN, access point (AP) is split into two pieces: wireless termination point (WTP), and access controller (AC). An AC is the centralized controller for a large number of WTPs. A station (STA) has the demand to move from one WTP to another within the same extended service set (ESS). Such behavior is called a basic service set (BSS) transition. In this paper, we propose the cluster‐chain‐based context (C3T) mechanism with a cluster member selection (CMS) algorithm to reduce the BSS transition delay in the centralized WLAN architecture. The proposed mechanism can have lower signaling cost for context transfer and does not have the ping‐pong problem comparing with the existing context transfer schemes in terms of BSS transition, such as the proactive neighbor caching (PNC) scheme, the configuration and provisioning for wireless APs handoff protocol (CAPWAPHP), and so on. Furthermore, since the original 802.11r fast BSS transition does not specify how to work in the centralized WLAN architecture, we incorporate the C3T mechanism into IEEE 802.11r that works in the centralized WLAN architecture. Copyright © 2008 John Wiley & Sons, Ltd.

A station (STA) has the demand of the basic service set (BSS) transition in the emergence of centralized wireless LAN (WLAN)architecture. The centralized WLAN architecture is a centralized hierarchical network and is suitable for large scale deployment. In this paper, we propose a context transfer mechanism using cluster-chain to reduce the BSS transition delay in the centralized WLAN architecture. Comparing with the existing context transfer schemes in terms of BSS transition, the proposed mechanism can have (i) lower signaling cost for context transfer and (ii) higher hit rate in term of STA's context.

For a ubiquitous multimedia streaming, an adaptive multimedia streaming scheme with multiple layered audiovisual (AV) coding is required over heterogeneous networks. This paper proposes a multiple layered audiovisual streaming scheme denoted ML-AVSS to deliver layered audiovisual data. Fine-granular scalability (FGS) and bit sliced arithmetic coding (BSAC) techniques are used to segment video and audio data into individual base-layer and multiple enhancement-layer bitstreams. With advantages of layered coding, two phases of streaming synchronization, including a human-perceptual based transmission scheme, de-jitter procedure, conditional retransmission and play- out synchronization, are proposed to transmit hybrid multiple layered audiovisual bitstreams. Experiment results show that the proposed ML-AVSS is a feasible streaming scheme to overcome challenges of the ubiquitous multimedia streaming, e.g., constrained available bandwidth, quality degradation, unsmooth playback, etc.

Energy conservation and network performance are the critical issues in wireless sensor networks (WSNs). This study designed and developed a novel communication efficient topology control protocol, which calls Load Sharing Topology Control Protocol (LS-TCP). LS-TCP is a well-designed protocol, including selecting suitable communication nodes, adjusting service loads of critical nodes, and adaptively sleeping management. LS-TCP is suitable for harsh environments without sensor identifications (ID) and locations. The proposed protocol principally divides the topology operation into topology formation phase and topology adjustment phases. In the topology formation phase, a enhance tree are set up. In the topology adjustment phase, enhance tree is adjusted with an optimal balance of critical nodes in backbone. Simulation results are performed using LS-TCP, which efficiently achieve much longer system life time in harsh environments.

In this paper, we proposed a partially reliable - concurrent multipath transfer (PR-CMT) that is featured with partially reliable multi-path transmission based on the concurrent multipath transfer (CMT) and the stream control transmission protocol partial reliability extension (PR-SCTP). The proposed partially reliable transmission is realized by the "timed reliable service". Data in PR-CMT are associated with lifetime. When the lifetime of the data is expired, PR-CMT would not transmit and retransmit the expired data. The false acked TSNs problem is identified in PR-SCTP and CMT. In this paper, the proposed PR-CMT can adopt some mechanisms to eliminate the false acked transmission sequence numbers (TSNs) problem. Simulation results show that PR-CMT can (i) have better transmission efficiency and (ii) consume less bandwidth.

Abstract The radical development in information technology makes the information appliances (IAs) able to interact and share information using the universal plug and play (UPnP) protocol. Inter‐IA interactions that involve more than one IA do exist in the future digital home network. Nowadays, inter‐IA interactions are developed by system and service providers and are fixed. It is difficult for users to customize their inter‐IA interactions. In this paper, we propose the service interaction script (SIS) language, which is a flow control language for users to customize their own inter‐IA interactions through the Web interfaces in the UPnP home network. Domain‐specific statements for the development of the inter‐IA interactions in SIS are (i) IA access statement, (ii) parallel statement and (iii) interaction event handling statement. On the other hand, the ubiquitous IAs access platform (UIAP) that allows users to use handheld devices to access the IA base on the UPnP protocol is also developed. UIAP provides the Web interfaces, i.e. (i) control panel and (ii) interaction customization panel, for users to access IAs remotely using handheld devices with or without the UPnP capability. The interaction customization panel contains (i) an easy mode and (ii) an advanced mode for users to customize inter‐IA interactions defined by themselves through the Web interfaces using SIS. In the easy mode, inter‐IA interaction customizations are achieved through the Web graphical user interface components and translated into SIS programs; in the advanced mode, users can write SIS programs directly. Copyright © 2007 John Wiley & Sons, Ltd.

In IP Multimedia Subsystem (IMS) of UMTS, both packet-switch domain and IMS authentications are necessary for the IMS subscriber, so-called the two-pass authentication. However, the IMS authentication is carried out by IMS authentication and key agreement (IMS AKA). Since IMS AKA is based on 3GPP AKA that is used to Packet-switch domain authentication, almost all of the operations are the same. It is inefficient that almost all involved steps in the two-pass authentication are duplicated. Hence, Lin et al.'s proposed the one-pass authentication to increase efficiency of the IMS authentication. Unfortunately, in addition to some security problems, the one-pass authentication procedure only has unilateral authentication without capabilities of mutual authentication and key agreement. Therefore, this paper proposes an one-pass AKA procedure, which not only avoids the lacks of the one-pass authentication procedure but also does not lose the efficiency.

One of the most critical disadvantages of the IP- based surveillance is the network reliability problem. That is, since the IP-based surveillance system is IP-based, all of the services provided by the IP-based surveillance system will not be accessible at all if network failure occurs. In this paper, a network-fault-tolerable IP-based surveillance (NeTSurv) system is proposed. In addition to the existing features of current video surveillance systems, a multihome-based approach is adopted to improve the network fault tolerance capability of the NeTSurv system. Furthermore, the fault tolerance mechanism is developed to solve the connectivity issue and the continuity issue of video playout for different link failure scenarios.

In order to adapt the quality of videos streamed over a time-varying bandwidth channel, a network-aware bandwidth management and rate control are required. This paper proposes the well-designed bandwidth estimation and active buffer control for streaming H.264 FGS videos over heterogeneous wired/ wireless/3G networks. According to the information of measured packet round-trip-time, loss-rate and delay jitter, an improved Kalman filter is proposed to predict the throughput recursively, and determine the transmission rate in consideration of buffer fullness of a decoder. The optimal parameters of the Kalman filter, e.g., transition matrix and error covariance, can be initialized, converged and adapted to the current network, even when the session handoff occurs. In our real experiments, distinct network traffic models are simulated, and corresponding estimation results w.r.t. network information are also exhibited.

In a Universal Mobile telecommunications system IP multimedia subsystem (IMS), even when an IMS subscriber has passed the packet-switch domain authentication, the IMS subscriber's identity must be confirmed by the IMS authentication again before accessing IMS services. Both the packet-switch domain and the IMS authentications are necessary for the IMS subscriber. This is referred to as a two-pass authentication. However, the packet-switch domain authentication is carried out by the authentication and key agreement (AKA) of the 3rd generation partnership projects (3GPP), called 3GPP AKA; the IMS authentication is carried out by IMS AKA. Since IMS AKA is based on 3GPP AKA, almost all of the operations are the same. It is inefficient because almost all the involved steps in the two-pass authentication are duplicated. Hence, the one-pass authentication was proposed to increase the efficiency of the IMS authentication. Unfortunately, the one-pass authentication has some security problems and loses the mutual AKA capabilities. Therefore, in this paper, an evolutionary IMS AKA (E-IMS AKA) in the IMS authentication is proposed to replace the IMS AKA. E-IMS AKA not only adheres to the security requirements of IMS AKA, but also maintains the efficiency of the one-pass authentication method.

The 3G mobile data network provides always-on and ubiquitous connectivity for subscribers. Although the service coverage area in wireless local area network (WLAN) is much smaller than that in a 3G mobile data network, the data transmission rate in WLAN can be from 2 to 54 Mbps, which is much faster than 3G mobile network. Obviously, the relationship between the 3G mobile data network and WLAN is complementary in terms of service coverage and data transmission rate. Therefore integration of 3G mobile network and WLAN can offer subscribers higher speed wireless service in hot spots and ubiquitous connectivity in 3G mobile data network. An authentication mechanism over the loose coupled integration mechanism using a cross-layer bootstrap is proposed. The benefits of the proposed mechanism are (a) integrating Universal Mobile Telecommunication System network and WLAN using the existing protocols defined in 3GPP, IETF and IEEE 802.11i, (b) the use of the Extension Authentication Protocol authentication method is flexible, (c) reduction of the authentication signalling when a subscriber roams from one access point (AP) to another AP and (d) user identity privacy protection.

With the coming wireless mobile networks era and the popular use of P2P applications, how to improve the resource retrieval and discovery for P2P file sharing applications in wireless mobile networks becomes a critical issue. In this paper, we propose a novel network-aware P2P file architecture and related control schemes that can provide continuous resource retrieval and discovery for mobile users over the wireless network environment. The proposed architecture divides a P2P file sharing network into multiple network-aware clusters, in which peers are assigned to a network-aware cluster using a network prefix division. Accordingly, there are two designs for supporting mobile peers to retrieve files in wireless mobile networks. First, a novel file discovery control scheme named mobility-aware file discovery control (MAFDC) scheme is devised to obtain fresh status of shared peers and find the new resource providing peers in wireless mobile networks. Second, a resource provider selection algorithm is devised to enable a mobile peer to select new resource providing peers for continuous file retrieval.

Proxies can improve the quality of service (QoS) of clients in the three-tier networking architecture. However, it is more complicated to apply the server-proxy-client networking architecture to mobile networks for multimedia streaming because mobile clients possess the "keeping moving" characteristic. Therefore, the three-tier architecture in mobile networks must take user mobility into consideration, i.e., mobile clients should be able to switch to a proxy dynamically. In this paper, application-layer proxy handoff (APH) is defined to have applications be executed smoothly when mobile clients move in the server-proxy-client architecture. First, APH employs application-layer anycast to select one of the candidate proxies as the next proxy based on 1) the network condition between the mobile client and each candidate proxy and 2) the load balance among the candidate proxies. Second, APH utilizes IPv6 multicast to switch the session from the original proxy to the next proxy smoothly and to forward the available cache unsent in the original proxy to the next proxy for keeping the original session continuous

Bit-rate control plays a major role In video coding and multimedia streaming. A well-designed bit-rate control mechanism can achieve line visual qualities and avoid network congestion over a time-varying channel. This paper proposes an H.264 bit-rate control using a 4D perceptual quantization modeling (PQrc), including two major encoding modules: the perceptual frame-level bit-allocation using a 1D temporal pattern and the macroblock-level quantizer decision using a 3D rate pattern. The temporal pattern is used to predict frame complexity and determine proper budget bits further. The rate pattern is depicted as a bit-complexity-quantization (B.C.Q.) model, in which a tangent slope of a B.C.Q. curve is a piece of unique information to find a proper quantizer. For newly generated video clips, the B.C.Q. model is updated continuously using a weighted least-square estimation. In comparison with the latest H.264 JM10.2, our experiment results show that the proposed PQrc can: 1) keep stable buffer fullness and 2) improve the SNR quality and control accuracy effectively.

Mobile IP is a popular referenced mobile wireless protocol that provides the rules for mobile node to roam between networks. Although many researches have been proposed for mobile IP related development, the disruption network supporting time called handoff latency is still a challenging problem for roaming. In this paper, we state a link layer assisted fast handoff scheme using the alternative path (LLA-AP) approach to improve performance of roaming in the wireless mobile network. In the proposed LLA-AP scheme: (i) signal strength information, i.e., link layer (L2) information, is provided to measure the better path decision and (ii) multiple interfaces are equipped with each mobile node to switch to the better path in advance. For the performance analysis aspect, we also construct a practical testing environment to verify the effectiveness of our proposed LLA-AP scheme.

Bit-rate control has a critical influence in video coding and multimedia streaming. This paper proposes a novel H.264 bit-rate control using a 3D perceptual quantization modeling (PQrc), including two major encoding modules: the perceptual frame-level bit-allocation and the fast macroblock-level quantizer decision. The frame-level budget bit depends on the frame complexity (isin) and buffer fullness, in which e is weighted by the predicted mean-absolute-difference (MAD) and the just-noticeable-difference (JND) PSNR. Considering the MB- level quantizer decision, the 3-D bits-complexity-quantization, which is denoted as B.C.Q., model is established, in which the B.C.Q. curve's tangent slope is a piece of unique information to find a proper quantizer. In comparison with the latest H.264 JM10.2, our experiment results show that the proposedPQrccan improve the SNR quality and keep the stable buffer fullness with less computational cost.

The movement of an entire mobile network is so-called "Network Mobility". Network mobility can widely apply to public transport, such as WLAN aboard. The IETF has proposed the NEMO basic support protocol to handle the network mobility problem. However, the NEMO basic support protocol results in sub-optimality that may not be tolerated by multimedia streaming services. In this paper, a framework, called SIP-NEMO, extends SIP to support network mobility. The proposed SIP-NEMO achieves route optimization not only when the caller and the callee are in different networks but also when the caller and callee are inside the same mobile network. Moreover, in order to save the bandwidth of wireless links, SIP-NEMO reduces the signaling messages using a URI-list service.

Video streaming hot-plug is defined as keeping video streaming to be continuous when a host is switched from wired/wireless to wireless/wired networks, or from one wireless network to another wireless network. In this paper, video streaming hot-plug mechanisms using Stream Control Transmission Protocol (SCTP), which is a new transport layer protocol supporting multihoming, are investigated. Using SCTP, a host can establish multiple paths, e.g. one through the wired network and another one through the wireless network, to communicate with its peer. Two SCTP extensions, the SCTP ADDIP extension and (Partially Reliable) PR-SCTP are adopted to implement the corresponding system called MOVIDEO. The SCTP ADDIP extension enables the host to dynamically configure IP addresses and the primary path used for the association, whereas PR-SCTP provides a partially reliable transmission option for the host to transmit data. Performance evaluation of the MOVIDEO system is provided in this paper. In addition to MOVIDEO, three other configurations of SCTP, including the triggering patterns of changing primary path and the transmission patterns of streaming video data, are implemented for performance comparison. The experimental results demonstrate that, using the proposed SCTP-based video streaming hot-plug mechanism, the connection of the video streaming service would not be broken and the playout of the video can be kept smooth during the process of video streaming hot-plug.

With the provision of various wireless services, e.g., third-generation (3G) and wireless local area network (WLAN), more and more people request to access the Internet anywhere at anytime. For example, people want to check their e-mails on the bus or watch online news while traveling in the train. For this purpose, the Internet Engineering Task Force (IETF) proposed the concept of network mobility, i.e., a set of users move as a unit. We motivate the network mobility problem by considering the state-of-the-art scenario of the network mobility (NEMO) basic support protocol that is extended from the Mobile IPv6 (MIPv6). In order to avoid the same problems suffered by MIPv6, a new session initiation protocol (SIP)-based network mobility management scheme called SIP-NEMO is designed and proposed in this paper. The proposed SIP-NEMO not only copes with the movement of a mobile network but also achieves the route optimization between two SIP clients without too many signaling messages over wireless links, even if the mobile network is nested. In this paper, we also analytically compute and simulate the performance of SIP-NEMO with the NEMO basic support protocol proposed by the IETF.

This paper presents our efforts on implementing an all-lPv6 SOHO (small office and home office) router for network users to connect all-IPv6 networks easily. Our proposed SOHO router is developed on embedded Linux system, which model is ADM 5120 using Linux-kernel 2.4.18 that supports IPv6 and produced by Infineon-ADMTek Inc., it can satisfy the four basic demands of a generic SOHO router, such as auto-configuration for network setting, packet routing for network traffic, packet filtering for network security, and small-sized and low-priced demands.

Radio interface and wireless service accesses are two areas, in which wireless networks do not provide the same level of protection as wired networks. It may result in the wireless networks being vulnerable to various attacks. For example, the first generation cellular mobile communication systems, such as advanced mobile phone service (AMPS), and the second generation cellular mobile communication systems, such as global system for mobile (GSM), do not take account of security issues comprehensively. It brings about an intruder to be able to eavesdrop user traffic or even change the identities of mobile phones to gain fraudulent services. Therefore, the Universal Mobile Telecommunication System (UMTS), which is known as the third generation (3G) cellular mobile communication system, is proposed. UMTS adopts the security feature of GSM in order to interwork with GSM smoothly. Furthermore, the UMTS adds new security features to design an authentication and key agreement protocol, which is called UMTS AKA protocol. However, it is found that the UMTS AKA has some problems, including bandwidth consumption between a serving network and a user's home network, space overhead of the serving network and sequence number synchronization. Therefore, in this paper, we propose an extension of UMTS AKA protocol, called UMTS X-AKA to overcome the aforementioned problems of UMTS AKA.

A peer-to-peer (P2P) file sharing network provides a resource sharing platform for Internet users. To increase higher-degreere source sharing, heterogeneous P2P file sharing networks need away to collaborate and communicate with each other. Based on the approach of interconnecting heterogeneous P2P file sharing networks, users on one P2P file sharing network can share resources and search data with other P2P file sharing networks. The main objective of our work is to enable connectivity and universal access among heterogeneous P2P file sharing networks. A novel architecture for P2P file sharing network interconnection called Shoran is proposed to serve peers' requests and route peers' requests to other P2P file sharing networks. Shoran provides (i) a message routing mechanism to route query messages among different P2P file sharing networks and (ii) a uniform message format that can ease the message exchange among heterogeneous P2P file sharing networks.

We proposed a ubiquitous 1-to-k media streaming architecture in which IPv4 and IPv6 clients in the transitive environment can be served simultaneously. The architecture is achieved by the adoption of multicast transition gateway (MTG). The deployment of MTGs in the architecture induces the requirement of efficiently selecting a proper MTG for streaming relay. An MTG selection framework is also designed for this issue.

In a server-proxy-client 3-tier architecture of the wireless mobile network, proxies should be dynamically assigned to serve mobile hosts according to geographical dependency and the network situation. The goal of proxy handoff is to allow a mobile host still can receive packets from the corresponding server even though its served proxy is switched from the current one to the other one. A 3-tier multimedia mobile transmission platform, called MMTP, is proposed to solve the proxy handoff problem in the IPv6-based wireless mobile network. In MMTP, a proxy handoff scheme, which is based on the application layer anycast technique, is proposed to select from a number of candidate proxies according to the load balance concern and wireless network situation concern.

This paper investigates the implementation of video servers employing multi-zone disks. A multi-zone disk consists of a set of concentric disk zones with non-uniform storage capacities and data transfer rates. Superimposing a customized region layout on the top of such a disk is essential for facilitating the video server design and improving the disk resource utilization. This paper develops theories concerning the optimal region layouts for implementing the region-based data placement. Based on them, two efficient region layout algorithms are proposed, namely, the free-π scheme and fixed-π scheme. The former can achieve better resource utilization than the latter, but the latter requires a simpler timing control than the former. Extension of the fixed-π scheme to perform the disk striping in a disk array platform is also investigated. A spiral synchronization scheme is proposed to improve the system scalability by alleviating the disk bus congestion problem, caused by variable data transfer rates from individual disks. The analytical results reveal that the proposed schemes can effectively customize region layouts to achieve 85–99% resource utilization. By considering the region layout together with other design factors, an efficient configuration strategy is proposed for building a cost-effective server subject to the performance requirements of a target application.

The provision of many VCR-like user interactions, e.g. reverse, skip, freeze-restart and scale, makes multimedia applications more flexible and useful to users. However, due to the unpredictable and uncontrollable network delay and jitter, the provision of VCR-like functions complicates the processing of multimedia synchronization a lot. The main reason is that (i) those media units stored in the client buffers and transmitting in the networks may be either useful or useless and (ii) the servers ’ media transmission should be adjusted when the corresponding presentation is resumed. Additionally, different VCR-like user interactions have their own processing considerations, which require different control schemes. In this paper, we propose some control schemes that are based on the feedback-adaptive mechanism. Based on the control schemes, we developed a Multimediaon-Demand system in which text, image, graphics, audio and video media are transmitted from the server site to the client site through text, image, graphics, audio and video communication channels respectively. Some performance measurements of the proposed control schemes are also given in this paper. 1.

Real time video streaming over wireless networks is not trivial because of the two characteristics of the wireless networks - (1) smaller bandwidth and (2) unreliable transmission media. An adaptive real-time video streaming scheme that uses the layered video technique is proposed. Two attributes that are used to determine the network situation and then adjust the sending rate accordingly are loss rate and round trip time (RTT). Since the unreliable media cause of packet loss in the wireless environment is rate-independent, a method that can separate the rate-independent loss from the congestion loss is needed. We use inter-arrival time between two received packets to identify if an out of order packet was received in time to do the packet loss separating. When a congestion situation is determined, the sending rate of the sender is dropped down one layer; when an unloaded situation is determined, the sending rate of the sender is raised up one layer. In this way, the adaptive real-time video streaming can be achieved for wireless networks.

In this paper, we proposed a DiffServ based multicast media gateway, which is called adaptive differentiated service multicast gateway (ADMG), to achieve the QoS guarantee and scalability considerations for multimedia presentation systems. The purpose of the ADMG system is twofold: (1) improve the bandwidth utilization by doing bandwidth reservation based on the layered-media aggregation and (2) improve the receiving quality of the layered media streams while the network is under heavy traffic by dynamically adjusting the sending rate according to the networking situation.

In recent years, some methods have been proposed to estimate null values in relational database systems. However, the estimated accuracy of the existing methods are not good enough. In this paper, we present a new method to generate weighted fuzzy rules from relational database systems for estimating null values using genetic algorithms (GAs), where the attributes appearing in the antecedent part of generated fuzzy rules have different weights. After a predefined number of evolutions of the GA, the best chromosome contains the optimal weights of the attributes, and they can be translated into a set of rules to be used for estimating null values. The proposed method can get a higher average estimated accuracy rate than the methods we presented in two previous papers.

New global trends in communications technology and liberalization are forcing dramatic changes in our lives. Applications tend to be lightweight and mobile, resulting in greater convenience. We present a new architecture to advance the traditional authoring system towards these trends. In the proposed architecture, the authoring system is capable of editing media content distributed in the networks. It forms a new model of authoring, i.e., the distributed authoring model. The distributed authoring model is described and compared with the traditional authoring model. We introduce quality-on-demand streaming in the architecture for the purpose of distributed authoring. Furthermore, we address the problems of applying multiple source streaming and quality adaptation in the architecture and show our design for resolving these problems. An implementation conforming to this architecture is also described.

Exploitation of the statistical multiplexing gain in the delivery of compressed videos is essential for achieving efficient network resource utilization. This paper proposes an analytical framework based on a GOP-aware effective envelope for guaranteeing statistical quality-of-service (QoS) to an aggregate flow from a class of connections. The GOP-aware effective envelope intends to model more realistic traffic burst by taking the information of individual flow's launch time and group-of-picture (GOP) into account With this information, the QoS guarantee is applicable not only to the aggregate flow but also to individual flows. Through numerical experiments, we show that the significant improvement on multiplexing gain can be achieved, particularly, under those cases with short delay bounds and small-to-moderate aggregation scales.

The demand for a faster, more efficient and more convenient way for accessing the WWW is urgently required with the growing use of the Internet. Thus, even for many non-computer devices, such as telephones, fax machines and hand-held devices, there have been added some new networking features and corresponding new applications created. Different I/O devices have different characteristics and browsing behaviors. To provide a versatile service to different I/O devices with different quality of information, we propose a mark-up language called the Quality of Information Mark-up Language (QIML( and corresponding browsing systems in this paper. With QIML, a page designer can describe a different quality of information for presentations to different I/O devices using a single Web page set. The language structure, system architecture and system development are presented in the paper.

A media filter can ensure full quality media playout at the high-performance computing and communicating environment while lowering to an appropriate media quality for the smooth playout in a limited bandwidth network. We present a new streaming architecture for the establishment of continuous media data flow over heterogeneous computer networks. Based on the new streaming architecture, we have developed a flexible and efficient audio/video transmission system. In order to eliminate heterogeneity problems found in networks, end-systems and applications, we adopt the concept of QoS media filter. The individual or receiver-driven quality of service requirement can be achieved through the application of a QoS media filtering mechanism.

In the paper, we propose the pause-and-run approach for k-stream (PARK) multimedia presentations over the Internet to achieve reliable transmission of continuous media. The main characteristics of the PARK approach are as follows: (i) to achieve reliable transmission of continuous media, PARK adopts TCP instead of UDP; (ii) since the slow-start scheme is embedded in TCP, a novel flow adaptation scheme is proposed to reduce the overhead of the network and end hosts. The server adapts its transmission rates to the buffer situation of the client and prevents the client's buffers from overflow and underflow as much as possible; (iii) with the provision of multiple-stream synchronization and multi-level adaptation control, the client achieves smooth multimedia presentations and achieves graceful presentation degradation when the resources are insufficient.

Complete verification of communication protocols is usually very hard to achieve due to the combinatorial state space explosion problem. Probability based partial verification provides an alternative solution to solve this problem. We adopt a Timed Communicating State Machine (TCSM), which belongs to the Extended Communicating Finite State Machine (ECFSM) model, to formally specify protocols that incorporate timed properties as part of their specifications. Based on the TCSM model, we propose a probabilistic timed verification scheme that is based on the occurrence rates of communicating entities' transitions and occurrence probabilities of channel entities' transitions. Using our probabilistic partial timed protocol verification scheme, an Estelle based Probabilistic Partial Timed Protocol verification system, which is called PTPVS, is developed on SUN SPARC workstations. In this way, protocol designers can use PTPVS to design and partially verify Estelle based protocol specifications with time properties.

Synchronization control with VCR-like user interactions in the multi-stream environment is very complicated and difficult. The main reasons are that: (i) those media units stored in the client buffers and flowing in the networks may be either useful or useless when the user interaction is issued, and (ii) the servers' media transmission should be adjusted according to the presentation status of the client. Additionally, different VCR-like user interactions have their own processing considerations, which result in different control schemes for different VCR-like user interactions. We propose some control schemes to handle interactive multimedia communication in the distributed multi-stream environment and develop an MOD system on SUN SPARC workstations accordingly.

This work presents the development of an integrated computer and telephone-accessed WWW system (CTW) to provide a ubiquitous web access service. The CTW system serves as an intermediary between the telephone user and web sites. The CTW system retrieves and then converts the proposed Hyper Phone Markup Language (HPML) specified web pages into speech patterns. Telephone users can access information, including form-based transaction services, on HPML-specified web pages using a telephone set anywhere in the world by connecting to the CTW system. Another feature of CTW is that HPML- specified web pages stored in cyberspace can also be accessed via regular browsers, including the current Netscape Navigator and Microsoft Internet Explorer. The CTW system provides a ubiquitous web access service for users at any degree of web-literacy by integrating well-installed telephone networks and the Internet. Copyright © 2000 John Wiley & Sons, Ltd.

The characteristics of multicast communication and delivery of multiple media streams that exist in a distributed multimedia system induce many new problems. Two main problems are (i) multiple media streams, in which of them is multicasted via its dedicated transmission channel, make the synchronization problem complicated; (ii) end-users that are scattered on different LANs may have different network situations and may have different presentation considerations. To resolve the above two problems, a presentation system called Mcast that can achieve smooth multiple-stream multimedia presentations for heterogeneous clients is proposed and developed. In this paper, system modules of Mcast, including multicast communication architecture, and synchronization and presentation mechanisms are described. With the detailed description of system modules, this paper provides a feasible guideline for developers of multicast multiple-stream multimedia presentations.

Due to the widespread distribution and the nature of hyperlinks on the WWW system, interactive temporal behaviours that exist in the WWW environment are different from those used in traditionally distributed multimedia presentation environments. Therefore, the control scheme and model of a new temporal synchronization that can guarantee the presentation continuity within the multimedia WWW environment is required. In this paper we will: (i) classify interactive temporal behaviours that are associated with user interactions, i.e. `replay', `freeze-restart', `skip&go-to', `hold&go-to' and `end&go-to', into two classes, intrapage user and interpage interactive temporal behaviours; (ii) identify the link control with some temporal constraints; (iii) derive temporal interactive synchronization control schemes to resolve the permitted interactive temporal behaviours and the system's link behaviour; and (iv) propose the extended finite state machine (EFSM) approach that is able to formally model the corresponding interactive temporal and link control architecture and schemes.

Many networking applications become feasible and required with the popular use of Internet. We present an Interactive Digital Radio Station, which is called IDRS, over Internet. Several architecture and synchronization methods are proposed to solve the communication problems that only occur during the digital radio multicasting. In order to perform the communication between DJ and audience, two operation modes, which are called the DJ mode and the Call-in mode, are defined. Related technique issues and the corresponding system development are presented in detail.

To verify the logical correctness of communication protocols, protocol verification should be invoked in the design phase. In this paper, we propose a new communicating finite state machine (CFSM)-based verification method, which is called reverse protocol verification. By analyzing the properties of deadlock error, unspecified reception error and channel overflow error, some candidate erroneous global states are generated. Then, each candidate global state is checked whether there is a path, i.e. a global state sequence, which can connect to the original initial global state. If there is a path, then the candidate global state is really an erroneous global state and the protocol under designing does have some logical errors; alternatively, if there is no candidate global state or none of the candidate global states has a path, then the protocol is error free. Based on the reverse method, an incremental protocol specification system (IPSS) has been developed on SUN SPARC workstations. Usage and applications of IPSS are also presented in this paper.

Multimedia synchronization is one of the key issues to be resolved in order to have smooth multimedia presentations. The demand of bringing multimedia information systems into distributed environments makes multimedia synchronization more difficult. We apply extended finite state machines (EFSM) to formally specify multimedia synchronization among multiple media streams. Using the EFSM model, the intra-medium synchronization is handled by an actor, which is formally represented as an EFSM. The inter-media synchronization is handled by a synchronizer, which is also represented as an EFSM. Both centralized and distributed synchronization cases can be specified. The synchronizer and actors achieve the corresponding multimedia presentations cooperatively. We also propose some policies to rectify the random delays caused by networks in the distributed presentation case. These policies, which can be formally specified in the corresponding EFSMs, are classified as intra-stream and inter-stream synchronization policies. Using the EFSM-based model, an authoring system and an associated experimental execution environment has been developed on SUN SPARC workstations.

A multimedia document is composed of different media objects. ISO's Open Document Architecture (ODA) proposes a standard multimedia document model. However, the current ODA profile only includes static media, e.g. text, geometric graphics and images. Because the future multimedia documents not only include static media but also continuous media, e.g. video and audio, continuous media document parts should be added to have a complete multimedia document model. In this paper, we propose a multimedia document model, which is derived from ODA's concept. The proposed model is based on the object-oriented approach. Objects in the proposed document model are divided into two types: data objects and pseudo objects. Data objects are data structures of a document; pseudo objects are used to manage data objects. Based on the proposed model, a multimedia document authoring and presenting system (MMDS) is also developed on SUN SPARC workstations using the Solaris 2.X operating system

The demand of bringing multimedia information systems into distributed environments makes multimedia synchronization more difficult. In order to eliminate the side effects which result from delay jitters, we propose a bounded buffer allocation scheme, in which the audio stream adopts the blocking synchronization scheme and the video stream adopts the non-blocking synchronization scheme, for live audio and video presentations in this paper. The forward synchronization schemes are performed to overcome the asynchrony anomalies. Once some anomalies of presentations are detected, a forward re-synchronization scheme is triggered to eliminate the asynchrony anomalies. Neither a global clock nor a feedback mechanism is needed using the proposed method. Based on the proposed method, trade-offs between the presentation qualities and networking resources are mathematically calculated. According to these calculable trade-offs, users can derive their own (acceptable) presentation qualities of live video and live audio media based on their available networking resources.

We propose a Timed Communicating State Machine (TCSM), which belongs to the extended state transition model, to formally specify protocols that incorporate timed properties as part of their specifications. Based on the TCSM model we also propose (1) a timed global state reachability analysis that takes time bounds and predicates into consideration, and (2) a probabilistic timed verification scheme that is based on the occurrence rates of communicating entities' transitions and the occurrence probabilities of channel entities' transitions. In this way, probability-based partial timed verification can be achieved for extended-state-transition-specified timed protocols.

Protocol verification is an activity to assure the correctness of communication protocols. Global state reachability analysis is one of the most straightforward and easily automated protocol verification methods. This paper proposes an incremental protocol verification method for the Extended Communicating Finite State Machine (ECFSM) model. Incremental protocol verification allows protocols to be modified at the rune time of global state reachability analysis. Then, instead of re-exploring the modified protocols from scratch, global state reachability analysis is continued incrementally at the modification point. To enhance the efficiency, the proposed method incorporates the dead and live variables concept that is used in Chu and Liu's global state reduction technique (Chu and Liu, 1989). Using the proposed incremental protocol verification method, incremental protocol design environments are achievable for ECFSM-based Formal Description Techniques (FDTs), e.g. ISO's Estelle. Our application of the proposed method to Estelle is also briefly introduced in this paper.

Abstract A lot of current e‐mail systems are ASCII text‐based, e.g. the e‐mail system in Internet. In the coming era of multimedia, e‐mails will be composed of other non‐text‐based information. This paper presents an evolution approach to enhance the capabilities of the current text‐based e‐mail systems. Using the proposed approach, an adapter is added to the top of the current system. As a result, multimedia e‐mails can be transmitted without changing the kernel part of the existing systems. We also propose an encoding scheme that has a 7/8 bandwidth utilization rate. Using this encoding scheme, all of the non‐ASCII data formats, e.g. image, voice, graphics, and so on, will be transformed to the ASCII data formats. Recently, we have developed an experimental multimedia e‐mail system using the proposed method. Our system has been developed on SUN SPARC workstations and an X‐window‐based user‐interface is provided in the system.

The extended communication finite state machine (ECFSM) model, which belongs to the state transition model, has been used to formally specify protocols with context variables and predicates. Global state reachability analysis is one of the most straight-forward ways to verify communication protocols specified in the state transition model. Many CFSM-based global state reduction techniques have been proposed to reduce the complexity of global state reachability analysis. However, these reduction techniques cannot be directly applied to ISO's Estelle and CCITT's SDL, which are ECFSM-based formal description techniques (FDTs)-based protocol verification systems. Based on Itoh and Ichikawa's (1983) CFSM-based, Chu and Liu's (1989) ECFSM-based reduction techniques, and Huang et al.'s (1990) CFSM-based incremental verification technique, this paper proposes a protocol verification technique for ECFSM-based n-entity protocols. In this way, the integrated reduced incremental verification technique can be directly applied to Estelle or SDL -based protocol verification systems.>

Formal description techniques (FDTs) provide formal and abstract ways to specify what protocols have to do and what the features of the protocols need. Protocol verification is a process for detecting logical errors, such as deadlock, unspecified receptions, and channel overflow errors, in communication protocols. Estelle is a FDT defined by the International Organization for Standardization (ISO) for protocol specifications. The authors present an incremental protocol verification technique and an Estelle interpreter which accepts Estelle protocol specifications for incremental protocol verification.>

A computer-aided protocol design system based on the OPS5 production system approach is proposed. Communication rules (state transitions) are specified as triples of object-attribute-value and the modeling of state transitions is specified by production rules. An incremental validation algorithm to facilitate protocol design is also presented. Based on a globally shared dataspace (working memory) in which different types and levels of information are all represented in a uniform structure, the OPS5 production system integrates both rule-based and procedure-based computations. Using this characteristic, the machine-dependent part can be abstractly specified through external procedure calls, the details of which can be coded in a procedure language until the implementation phase. Since computations in OPS5 are based on pattern matching, all of the attributes of elements that are in production rules or in external procedure calls can act as data templates for generic data types. This capability enhances the generic specification that allows different realizations for various implementation environments.>