Lin-Huang Chang

True suffix tree approach for discovering non-trivial repeating patterns in a music object

As the growing in Internet, database types and sizes are getting various and larger. The topic of finding out the significant information from a database at the shortest time is important. In the music databases, a repeating pattern is an important feature of music objects, which commonly used in analyzing the repeated part of music data and looking for themes. Most of the repeating patterns are key melodies or easy to familiarize and remember for people. Therefore, we can use the themes or the repeating patterns to construct indices that can speedup query execution for music retrievals. Nevertheless, non-trivial repeating patterns exclude those patterns, which are all contained in other longer patterns, such that they can reduce the redundancy of the repeating patterns and save the index space needed. Most of existing algorithms are time consuming for finding non-trivial repeating patterns in a music object. In this research, we aim to apply the true suffix tree approach to discover non-trivial repeating patterns for a music object, which can efficiently address the cost problems in processing time and memory space. In general case, our proposed scheme can extract non-trivial repeating patterns in a linear time.

QoS-aware path switching for VoIP traffic using SCTP

Voice over Internet protocol (VoIP) has been a prevalent multimedia service nowadays. It allows us to transmit voice data over IP networks. However, quality of service (QoS) is a major challenge to VoIP services. It must provide similar quality to traditional public switched telephone network or cellular phone services. Therefore, QoS related protocols have become important for real-time applications. Multi-protocol label switch (MPLS) is one of the important techniques to improve the network performance from QoS point of view. It employs label swapping to speed up packet forwarding. However, when a large number of users utilize VoIP services, the network congestion issue still exists. It causes delay, jitter and packet loss that affect VoIP QoS. In this paper, we propose a QoS-aware path switching strategy by using stream control transmission protocol (SCTP) in MPLS network to improve the VoIP traffic. This was done by employing SCTP selective acknowledgment mechanism to report the transmission parameters of primary path and to determine the criteria to switch to backup path. Simulation results show significant improvement in VoIP QoS.

Design and realization of ad-hoc VoIP with embedded p-SIP server

In this paper we design and implement the pseudo session initiation protocol (p-SIP) server embedded in each mobile node to provide the ad-hoc voice over Internet protocol (VoIP) services. The implemented p-SIP server, being compatible with common VoIP user agents, integrates the standard SIP protocol with SIP presence to handle SIP signaling and discovery mechanism in the ad-hoc VoIP networks. The ad-hoc VoIP signaling and voice traffic performances are analyzed using E-model R rating value for up to six hops in the implemented test-bed. We also conduct the interference experiments to imitate the practical ad-hoc VoIP environment. The analyzed results demonstrate the realization of ad-hoc VoIP services by using p-SIP server.

Dynamic load balancing for wired and wireless internet access

In this paper we propose a new algorithm, which analyzes arrival rate and link rate (or service rate) to fmd the weight for each of the outbound links automatically. Our purpose is to distribute traffic load evenly to each of the links. We also discuss the outbound links with wireless connections. The impact of retransmitting packets due to the high error probability on the traffic load will be considered in our system. The simulation results show that the proposed algorithm improves the performance for about ten to seventeen percentages.

A Handover Mechanism Using IEEE 802.21 in Heterogeneous Wireless Networks

The IEEE 802.21 framework is a novel standard and the scope of this standard is to optimize and facilitate handover in heterogeneous networks. In this paper, we propose a handover mobility management by using the IEEE 802.21 to solve some handover issues, such as handover delay and packet loss, in homogeneous and heterogeneous networks. By setting certain signal thresholds and probability confidence to trigger the Link Going Down event, the proposed handover mechanism is capable of providing better handover efficiency and reducing the handover delay and possible packet loss during handover. The NS-2 simulation is conducted to verify the performance of our handover mobility management across WLAN and WiMAX heterogeneous networks.

Design and implementation of the push-to-talk service in ad hoc VoIP network

Push-to-talk (PTT) is a walkie-talkie like service which performs an efficient and instant voice over Internet protocol (VoIP) communication in mobile ad hoc network especially for certain circumstances, such as battle field and earthquake or disaster relief. The authors have designed and implemented the PTT mechanism in ad hoc VoIP network. The PTT server and user agent combined with the pseudo session initiation protocol (SIP) server in the implementation provide the PTT service without standalone SIP server support. The authors also conduct the experimental measurements, in terms of delay and packet loss, in the test-bed to demonstrate the realisation of PTT service in ad hoc VoIP network.

A hierarchical care-of prefix with BUT scheme for nested mobile networks

In this paper, we apply the hierarchical concept to the Care-of Prefix (CoP) scheme as HCoP and enhance HCoP with a novel Binding Update Tree (BUT) structure as HCoP-B for NEtwork MObility (NEMO) management of the nested mobile network. As compared to schemes such as Reverse Routing Header (RRH), Route Optimization using Tree Information Option (ROTIO) and HCoP with numerical performance evaluations, HCoP-B achieves the shortest handoff latency and significantly reduces the consumed network bandwidth of global binding update messages for route optimizations (RO) of all correspondent nodes (CN) after the nested mobile network hands over to a new AR. Consequently, HCoP-B resolves the RO storm problem.

Mobility Management of VoIP Services Using SCTP Handoff Mechanism

Voice over Internet protocol (VoIP) is a prevalent multimedia application nowadays. For mobile nodes (MNs) using VoIP services in wireless networks, the handoff issues need to be taken into account. Stream control transmission protocol (SCTP) is a potential transport layer protocol with multi-homing and multi-streaming functions. Mobile SCTP with dynamic address reconfiguration (DAR) extends SCTP to mobile wireless network. In this paper, we design and implement a handoff mechanism using multi-homing, DAR and partially reliable SCTP (PR-SCTP) of mobile SCTP to transport VoIP applications. The switch from UDP or TCP to SCTP for VoIP transport provides the seamless handoff ability for MNs using real-time applications such as VoIP service. This paper also provides the performance analyses of the proposed scheme on the implemented Linux testbed.

Adaptive arrival rate dependent traffic balancing with redundancy

It is an important issue to provide the reliability as well as to balance the traffic load for the multihomed networks. In this paper, we deal with two most important factors, traffic load balancing and redundancy. The assigned weight for each link is dynamically computed according to the arrival and service rates via arrival rate dependent (ARD). By employing the adaptive-ARD (A-ARD) mechanism, the network performance in terms of average waiting time was further improved. The simulation result showed that more than 20% performance improvement was achieved by comparing A-ARD to RR, WRR and ARD with arrival rate 80 packets per second. Through the proposed redundancy scheme of A-ARD, the reliability of the packet transmission was fulfilled.

HMSIP: Hierarchical Multicast SIP Protocol for streaming multimedia in wireless networks

In this paper, we propose the Hierarchical Multicast Session Initiation Protocol (HMSIP), which integrates the IETF Session Initiation Protocol (SIP) with the IP multicast protocol and the RSVP resource reservation technique, to reduce the handoff delay, deployment cost and support fast handoff for the MH with guaranteed multimedia QoS that the traditional approaches such as the Hierarchical Remote Subscription (HRS) and Bi-directional Tunneling (BT) cannot do. We mathematically analyze handoff delays of the HRS, BT and HMSIP schemes. Finally, we show in simulations that the HMSIP can significantly reduce handoff delays suffered by the HRS and achieve more stable delays than the BT, with a small proportion of extra bandwidth than that of the bandwidth-optimal HRS.