Recently, the demand of broadband communications such as high-speed internet, HDTV, 3D-HDTV and ATM backbone network has been increased drastically. For transmitting the broad-bandwidth data using wireless network, it is needed to use ka-band frequency. However, the use of this ka-band frequency is seriously affected to the received data performance by rain fading and atmospheric propagation loss at the Ka-band satellite communication link. So, we need adaptive MODEM to endure the degraded performance by channel environment. In this paper, we will present the structure and design of the 155Mbps adaptive Modem adaptively compensated against channel environment. In order to compensate the rain attenuation over the ka-band wireless channel link, the adaptive coding schemes with variable coding rates and the multiple modulation schemes such as trellis coded 8-PSK, QPSK, and BPSK are adopted. And the blind demodulation scheme is proposed to demodulate without Information of modulation mode at the multi-mode demodulator, and the fast phase ambiguity resolving scheme is proposed. The design and simulation results of adaptive Modem by SPW model are provided. This 155Mbps adaptive MODEM was designed and implemented by single ASIC chip with the $0.25\mu{m}$ CMOS standard cell technology and 950 thousand gates.
Park, Jae-Won;Chung, Won-Young;Kim, Hyun-Pil;Lee, Jung-Hee;Lee, Yong-Surk
The Journal of Korean Institute of Communications and Information Sciences
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v.36
no.1B
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pp.80-85
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2011
The mount of network traffic from the Internet is increasing because of the use of Broadband Convergence Networks(BcN). Network traffic is also increasing because of the development of application, especially multimedia traffic from IPTV, VOD, and online games. This multimedia traffic not only has a huge payload but also should be considered a threat in real time. For this reason, this study examines the ways that routers distribute the bandwidth in accordance to traffic properties. To classify the property of the traffic, it is essential to analyze the application layer. However, the general network processor architecture serially processes the L2-4 and L7 layer. We propose a novel parallel network processor architecture with a global cache that processes L2-4 and L7 in parallel. To verify the proposed architecture, we simulated both of the architecture with SystemC. EEMBC and SNORT was used to measure L2-4 and L7 processing time. When multimedia traffic was entered into the network processor in the same flow, the proposed architecture showed about 85% higher performance than general architecture.
Proceedings of the Korean Institute of Surface Engineering Conference
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2018.06a
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pp.102-102
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2018
Electronic industry had required the finer size and the higher performance of the device. Therefore, 3-D die stacking technology such as TSV (through silicon via) and micro-bump had been used. Moreover, by the development of the 3-D die stacking technology, 3-D structure such as chip to chip (c2c) and chip to wafer (c2w) had become practicable. These technologies led to the appearance of HBM (high bandwidth memory). HBM was type of the memory, which is composed of several stacked layers of the memory chips. Each memory chips were connected by TSV and micro-bump. Thus, HBM had lower RC delay and higher performance of data processing than the conventional memory. Moreover, due to the development of the IT industry such as, AI (artificial intelligence), IOT (internet of things), and VR (virtual reality), the lower pitch size and the higher density were required to micro-electronics. Particularly, to obtain the fine pitch, some of the method such as copper pillar, nickel diffusion barrier, and tin-silver or tin-silver-copper based bump had been utillized. TCB (thermal compression bonding) and reflow process (thermal aging) were conventional method to bond between tin-silver or tin-silver-copper caps in the temperature range of 200 to 300 degrees. However, because of tin overflow which caused by higher operating temperature than melting point of Tin ($232^{\circ}C$), there would be the danger of bump bridge failure in fine-pitch bonding. Furthermore, regulating the phase of IMC (intermetallic compound) which was located between nickel diffusion barrier and bump, had a lot of problems. For example, an excess of kirkendall void which provides site of brittle fracture occurs at IMC layer after reflow process. The essential solution to reduce the difficulty of bump bonding process is copper to copper direct bonding below $300^{\circ}C$. In this study, in order to improve the problem of bump bonding process, copper to copper direct bonding was performed below $300^{\circ}C$. The driving force of bonding was the self-annealing properties of electrodeposited Cu with high defect density. The self-annealing property originated in high defect density and non-equilibrium grain boundaries at the triple junction. The electrodeposited Cu at high current density and low bath temperature was fabricated by electroplating on copper deposited silicon wafer. The copper-copper bonding experiments was conducted using thermal pressing machine. The condition of investigation such as thermal parameter and pressure parameter were varied to acquire proper bonded specimens. The bonded interface was characterized by SEM (scanning electron microscope) and OM (optical microscope). The density of grain boundary and defects were examined by TEM (transmission electron microscopy).
It is strongly believed that multicast will become one of the most promising services on internet for the next generation. Multicast service can be deployed either on network-layer or application-layer. IP multicast (network-layer multicast) is implemented by network nodes (i.e., routers) and avoids multiple copies of the same datagram on the same link. Despite the conceptual simplicity of IP multicast and its obvious benefits, it has not been widely deployed since there remain many unresolved issues. As an alternative to IP multicast, overlay multicast (application-layer multicast) implements the multicast functionality at end hosts rather than routers. This may require more overall bandwidth than IP multicast because duplicate packets travel the same physical links multiple times, but it provides an inexpensive, deployable method of providing point-to-multipoint group communication. In this paper we develop an efficient method applied greedy algorithm for solving two models of overlay multicast tree building problem that is aimed to construct MDST (Minimum Diameter Spanning Tree : minimum cost path from a source node to all its receivers) and MST (Minimum Spanning Tree : minimum total cost spanning all the members). We also simulate and analyze MDST and MST.
Mobile IPv6 spends considerable bandwidth considering that its signal volume is proportional to the mobile and also it should be strengthened to support the binding signal volume, the traffic, and effective mobility. So, the study in NEMO(Network Mobility), an extended version of Mobile IPv6, has been conducted. NEMO provides its mobility by putting several mobiles and more than one portable router into one unit called as mobile network. Because nodes access Internet via the portable router at this time, it receives transparency without any additional work and that much reduces binding signal while solving binding storm. By supporting mobility, NEMO is able to have various mobile structures which realize several networks hierarchically and it is necessary to improve its safety and security by authenticating among the upper networks or the lower ones while moving. Also, it is extremely required to begin a study in the device to improve efficiency accompanied with mobility, which is executed by the fast hand-off as well as the safe authentication. For those reasons, this paper not only classifies various NEMO mobile scenarios into 7 ways, but also provides AAA authentication of each scenario, the authentication through the safety authentication and fast handoff authentication using F+HMIPv6 and the way to reduce both signaling volume and packet delays efficiently during the handoff.
Wireless mesh networks can be deployed for various networks from home networking to last-mile broadband Internet access. Wireless mesh networks are composed of mesh routers and mesh clients. In these networks, static nodes form a multi-hop backbone of a large wireless access network that provides connectivity to end-users' mobile terminals. The network nodes cooperate with each other to relay data traffic to its destinations. In order to increase connectivity and better performance, researchers are getting interested in multi-channel and multi-interface wireless mesh networks. In these networks, non-overlapping multiple frequency channels are used simultaneously to increase the aggregate bandwidth available to end-users. Recently, researches have focused on finding suitable channel assignments for wireless network interfaces, equiped in a mesh node, together with efficient routing to improve overall system throughput in wireless mesh networks. This goal can be achieved by minimize channel interference. Less interference among using channels in a network guarantees more aggregated channel capacity and better connectivity of the networks. In this thesis, we propose interference aware channel assignment and routing algorithms for multi-channel multi-hop wireless mesh networks. We propose Channel Assignment and Routing algorithms using Traffic Profiles(CARTP) and Routing algorithms allowing detour routing(CARTP+2). Finally, we evaluate the performance of proposed algorithms in comparison to results from previous methods using ns-2 simulations. The simulation results show that our proposed algorithms can enhance the overall network performance in wireless mesh networks.
For playing a multimedia presentation in a Internet, the case that the presentation QoS(Quality of Services) at a destination nay be different from the QoS of multimedia data at the source occurs frequently. In this case, the process of trancoding the multimedia data at the source Into the multimedia data satisfying the QoS at the destination should be requited. In addition, even the presentation description having the homogeneous QoS at both sides may have different transcoding paths due to the limitation of display terminals or network bandwidth. That is, for a multimedia description, it is required to regenerate a proper transcoding path whenever the displaying terminals or the network environment gets decided. And the delay time required to go through the transcoding path may affect the playability of the give presentation. Therefore it should be checked whether the presentation requiring a transcoding process is able to be played in a real time. In this paper, the algorithm for generating all the possible transcoding paths for a given multimedia description under a fixed set of transcoders and the network environment is proposed. The algorithm adopts the concept of QoS transition diagram to Prevent from a trancoding Path being cycled by the repetition of a cyclic Path which generates the same QoS of multimedia data as its input QoS. By eliminating all the cyclic Paths, the algorithm can guarantee the termination of the process. And for the playability check, a method of computing the transcoding time and the delay lime between logical data units are proposed.Finally all the proposed methods were implemented in the stream engine, called TransCore and the presentation-authoring tool, called VIP, we had developed. And the test results with sample scenarios were presented at the last.
Recent portable devices are so versatile that they have multiple communication channels and play several multimedia formats. Especially, many services are under development for users who connect Internet or nearby devices via WWAN (Wireless Wide Area Network) and/or WLAN (Wireless LAN). In case of paying the telecommunication cost proportional to the amount of data downloaded, it is necessary to reduce the cost by constructing a special ad hoc network in which each participating peer downloads a specific portion of the want-to-be-shared content over the payable WWAN channel and exchanges the remaining portion with other peers using the cost-free WLAN channel. If all peers participate in forwarding packets, some transmissions are redundant which results in the unnecessary consumption of bandwidth as well as the delayed content distribution time. In order to reduce the redundant transmission, this paper proposes both the excluding method which discourages some peers not to forward redundant packets, and the minimum cover set method in which only the minimum number of peers are in charge of forwarding packets. These two methods obviate redundant packet forwarding, and result in reduction of content distribution time by up to around 29%.
The Journal of Korean Institute of Communications and Information Sciences
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v.31
no.8B
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pp.691-700
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2006
Based on the destination IP address of incoming packets, the Internet routers determine next hops and forward packets toward final destinations through If address lookup. The bandwidth of communication links increases exponentially fast as well as the routing table size grows significant as the number of single host networks attached to the Internet increases. Since packets should be processed at wire-speed, the increased link speed reduces the processing time of a packet in routers, and hence more efficient and fast IP address lookup algorithms and architectures are required in the next generation routers. Most of the previous IP lookup schemes compare routing prefixes of shorter length first with a given input IP address. Since IP address lookup needs to find the most specific route of the given input, search continues until the longest matched prefix is found while it keeps remembering the current test matching prefix. In this paper, based on binary search on prefix length, we proposed a new IP address lookup algorithm which compares longer prefixes first. The proposed scheme is consisted of multiple tries with prefixes on leaves only. The trie composed of the longest prefixes is primarily searched whether there is a match with the given input. This processing is repeated for the trio of the next longer prefixes until there finds a match. Hence the proposed algorithm provides the fast search speed. The proposed algorithm also provides the incremental update of prefixes while the previous binary search on length scheme does not provide the incremental update because of pre-processing requirement. In this paper, we performed extensive simulations and showed the performance comparisons with related works.
In recent years, the needs for WLANs(Wireless Local Area Networks) technology which can access to Internet anywhere have been dramatically increased particularly in SOHO(Small Office Home Office) and Hot Spot. However, unlike wired networks, there are some unique characteristics of wireless networks. These characteristics include the burst packet losses due to unreliable wireless channel. Note that burst packet losses, which occur when the distance between the wireless station and the AP(Access Point) increase or when obstacles move temporarily between the station and AP, are very frequent in 802.11 networks. Conversely, due to burst packet losses, the performance of 802.11 networks are not always as sufficient as the current application require, particularly when they use TCP at the transport layer. The high packet loss rate over wireless links can trigger unnecessary execution of TCP congestion control algorithm, resulting in performance degradation. In order to overcome the limitations of WLANs environment, MAC-layer LDA(Loss Differentiation Algorithm)has been proposed. MAC-layer LDA prevents TCP's timeout by increasing CRD(Consecutive Retry Duration) higher than burst packet loss duration. However, in the wireless channel with high packet loss rate, MAC-layer LDA does not work well because of two reason: (a) If the CRD is lower than burst packet loss duration due to the limited increase of retry limit, end-to-end performance is degraded. (b) energy of mobile device and bandwidth utilization in the wireless link are wasted unnecessarily by Reducing the drainage speed of the network buffer due to the increase of CRD. In this paper, we propose a new retransmission module based on Cross-layer approach, called BLD(Burst Loss Detection) module, to solve the limitation of previous link layer retransmission schemes. BLD module's algorithm is retransmission mechanism at IEEE 802.11 networks and performs retransmission based on the interaction between retransmission mechanisms of the MAC layer and TCP. From the simulation by using ns-2(Network Simulator), we could see more improved TCP throughput and energy efficiency with the proposed scheme than previous mechanisms.
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