A gasket system (18) is provided for use with a protective boot (12) and a pant leg (16) of a protective garment (14) for a firefighter or other emergency worker. The boot (12) includes an outer boot (24) and a moisture/vapor barrier liner (26), and the leg (16) includes an outer shell (20) and a moisture/vapor barrier liner (22). The gasket system (18) includes a first gasket (30) fixed to the liner (22) of the pant leg (16) and defining a first annular interlock (34) to encircle a wearer's leg received in the pant leg (16), and a second gasket (32) fixed to at least one of the outer boot (24) and the boot liner (26) and defining a second annular interlock (36) to encircle a wearer's leg received in the boot (12). The first and second interlocks (34,36) are engageable with each other to form a seal with a leg received in both the boot (12) and the pant leg (16).

Commonly, firefighters and other emergency rescue workers wear protective garments in which the distal margins of the pant legs of the protective garments may fit loosely over protective boots. Such loosely fitting margins do not effectively inhibit the ingress of potentially harmful materials, whether gaseous, liquid, or particulate. Several possible solutions to this challenge are shown in U.S. Pat. No. 6,134,717 to William L. Grilliot et al, issued Oct. 24, 2000, entitled "Protective Garment With Leg Portions and With Cuffs to Prevent Liquids from Entering Boots", U.S. Ser. No. 10/999,437 to William L. Grilliot et al, filed Nov. 30, 2004, entitled "Protective Garments Having Elastomeric Gaskets Along Margins to Inhibit Ingress of Potentially Harmful Materials" and U.S. Ser. No. 11/525,619 to William L. Grilliot et al, filed Sep. 22, 2006, entitled "Protective Garment Wearable with Boots and Comprising Attached Socks", the entire disclosures of which are incorporated herein by reference. While each of the foregoing may work well for their intended purpose, there is always room for improvement.

Protective garments and gasket system for firefighter's and other emergency personnel


In some embodiments, a method includes: generating entries for at least one cost plan, each of the entries including a planned amount and a descriptor, wherein the descriptor identifies a general ledger account. In some embodiments, a method includes: generating postings for at least one ledger, each of the postings including an actual amount, a first descriptor and a second descriptor, wherein the first descriptor identifies a general ledger account and wherein the second descriptor identifies a general ledger account.

A balance sheet is one type of financial statement that may be generated for a business. Some regulatory authorities require that a business generate a balance at least once per year. The regulatory authorities may also require that the balance sheet have a specific structure and/or specific accounts, sometimes referred to as balance sheet accounts. Examples of balance sheet accounts include, but are not limited to assets, liabilities, depreciation, etc. FIG. 1A is a schematic representation of a structure for a simple balance sheet 10. Some regulatory authorities may also assign a number and/or code to one or more of the required balance sheet accounts.
In a cost of sales method of accounting, direct costs attributed to the production of goods and/or services sold by a business are assigned to one or more balance sheet accounts. For example, direct costs associated with materials that have begun transformation to finished goods are assigned to a work in process (WIP) account 12 (FIG. 1A), which may be part of inventories in the assets portion of the balance sheet. Examples of goods and/or services that may be sold by a business include, but are not limited to, washing machines, cars and buildings. Notably, the goods and/or services that are sold by a business are sometimes referred to as cost objects. Manufacturing orders, sales orders, service orders and/or projects relating to goods and/or services are also sometimes referred to as cost objects.
Some businesses find that actual costs attributed to the production of goods and/or services may differ from planned costs for production of such goods and/or services. The difference between an actual cost and a planned cost is sometimes referred to as a variance.
Some businesses perform cost analysis to identify and/or help identify reason(s) for variances and implement changes to try to address such variances. For example, some businesses may make changes in regard to its choice of suppliers and/or processes used within the business. In some situations, the business may decide to change the selling price of the goods and/or services.
Various products are available to assist businesses in regard to accounting and/or cost analysis. Examples of such products include SAP-BW, and SAP-ERP component CO-PC, produced by SAP AG.
Notwithstanding the availability of such products, additional methods, apparatus, systems, computer program products and/or mediums to further assist in accounting and/or financial analysis are desired.

Methods, apparatus, systems, computer program products and mediums for use in association with accounting and/or financial analysis


A private mobile radio transmission system comprising a plurality of geographically spaced base stations (BS) defining a plurality of overlapping service areas (SA). The base stations are coupled for example by land-line communication links (CL) to a system controller (SC). Mobile stations (MS) equipped with simplex two-way radios are free to roam through the service areas. In order to be able to make a call with or via a base station it is necessary for the mobile station to be registered with the base station concerned. Details of the registration are stored by the mobile station concerned and the base station. In order to optimise on the usage of radio channels, details of a home base station are preprogrammed into each mobile station and when the mobile station wishes to register with a base station, it is instructed to try to register firstly with its own base station, on the basis that the majority of calls will be to and from the home service area and will require only one pair of frequencies, secondly with a base station of any other service area in its region, and thirdly with a base station of a service area in any other region. When registered with a service area other than its home one, the mobile station checks regularly to see if it is possible to register with its home service area.

The present invention relates to a mobile radio transmission system and to a mobile station for use in the system.
Such a system typically comprises a plurality of base stations situated at respective geographical locations to define a plurality of service areas within an overall region and a plurality of mobile stations capable of communicating by two-way radio with each base station when in communication range of that base station. Each base station comprises the entirety of transmitters and receivers operated by a trunking system controller (TSC) at any one site or service area. The communication link established is two-frequency simplex, that is one frequency is allocated to forward transmission from a base station to a mobile station and a second frequency is allocated for return transmission from a mobile station to a base station. Blocks of radio channels chosen from an overall band of channels are allocated to each base station. In the case of trunked radio, the base stations of a region may be connected by way of land line to a control station (CS) or may be connected as a mesh with the base stations interconnected with each other. There may be two or more regions in which case each region has its own CS which controls a plurality of associated base stations. The CSs are connected together by land lines.
In a known such system each mobile station is arranged when operative to register with a base station of which it is within communication range and thereby with the corresponding service area, and to respond to it being no longer within communication range of a base station serving a service area with which it is currently registered by registering with another base station of which it is within such range (if any) and thereby with the corresponding service area. The base station responds to the registration request by storing the identity of the service area corresponding to the relevant base station in that one of its location records which corresponds to the registering mobile station. The registration procedure is carried out so that if it is desired to call a particular mobile station it is known in which service area that mobile station is currently located; the CS arranges that the call is directed to the or each base station which serves the relevant service area, which base station in turn calls the relevant mobile station. An advantage of a registration procedure is that if there was no provision for registration then all the base stations of the system would have to call the particular mobile station, which would impose a considerable overhead on the transmission capacity of the system. However, a disadvantage with this known registration procedure is that if a mobile station is roaming near a border between two or more service areas so that it frequently moves between areas it will frequently change its registration from one service area to another, again imposing a considerable overhead on the communication channel capacity of the system.
European Patent Specification EP-A2-0260763 discloses a multiple registration method aimed at reducing the transmission overhead associated with the above described system. This multiple registration method enables the TSC to store a list of up to n registrations for each mobile. The mobiles each have storage facilities for storing the same list of registrations as are stored by the TSC in respect of itself. As a mobile enters a new service area it automatically registers with the area's base station thus updating the list of registration stored by the mobile and its associated TSC. By storing these registrations an enlarged service area is created. If a TSC wishes to contact a mobile it refers to the list of registrations which it is storing for that mobile and initially transmits a signal to the base stations which are registered, either successively, beginning with say the most recent registration, or simultaneously. If this is not successful then the TSC tries the other base stations. Provision is made for reordering the list on the basis that the channel number and frequencies of the service area in which a mobile last made a call are at the top of the list. In the event of the list of registrations not being updated and/or a call not being made to or received from a mobile after a set period then it is possible for registrations to be cancelled one at a time thereby reducing the size of the enlarged service area. While it is possible for all the registrations to be cancelled, it is preferred that at least one registration of a service area remain to prevent a flood of registrations at the start of a working day. If, at the start of a working day, a mobile is not registered in the service area in which it is currently located it has to arrange to immediately register. This method is quite workable but does have a drawback in that it will not necessarily optimise the usage of radio channels.
This drawback will be better understood by considering a hypothetical case of a circular geographical region of about 128 km with 7 base stations serving it. Obviously hilly areas tend to mask some mobiles from some base stations, and the signal strength obtained by a mobile from a base station varies inversely with the distance to the base station. The propagation map allows a generous margin of signal strength so that most mobiles can "see" most base stations.
The frequencies currently used in Great Britain for trunked private mobile radio systems are around 200 MHz, which are very penetrating and give good coverage, especially since the mobiles are extremely sensitive.
The power and number of radio channels fitted per base station are carefully chosen to correspond to the expected activity local to that base station.
The radio protocol used, for example a subset of MPT 1327-A Signalling Standard for Trunked Private Land Mobile Radio Systems issued by the Department of Trade and Industry, London 1988, effectively requires the mobiles to hold on to a registration as long as they can. The intention is, of course, to reduce the overhead of registration signalling on the network, which supplies no calls to the user.
Furthermore, the MPT 1327 protocol has a good error-correction layer, and so a significant number of on-air bits can be lost before any particular signalling packet fails. In consequence a mobile will not attempt to re-register on another base station unless the packet-level error rate on his present one is high. However on the fringe of the coverage area this can lead to a poor grade of service both in terms of speech signal-to-noise and in call set-up failure rate.
The result of the above set of circumstances is to change the radio topology of the network from being a number of largely isolated cells with little overlap, resulting in the registration map largely corresponding to the geographical position of the mobile, to one in which the cells very greatly overlap, causing the registration map to be largely random. Each base station has a number of mobiles registered on it proportional to its radio coverage, and these mobiles are distributed over the part of the region in which this can be "seen". Each mobile has only a small chance of being registered on the base station where it is most likely for its calls to be satisfied by that base station alone, which base station may be regarded as the mobile's home site or home service area.
The effect of this is that a high percentage of calls say 80% are intersite calls, using 2 radio traffic channels per call instead of one, the system capacity is almost halved, and the number of calls each base station must support bears little relation to the carefully estimated local population on which its size and power were based. This means that a base station gets a load proportional to the size of its radio coverage, not the number of channels fitted. This has the potential of grossly overloading the smaller service areas. Creating more service areas within the region by adding more base stations actually makes things worse, since it means that each mobile's chance of being registered on its home site is proportionally reduced.

Mobile radio transmission system using preferred home service area


There is provided a chair that realizes a state preferably following a motion of a sitter in accordance with a posture of the relevant sitter, and a state preferably supporting the sitter. The chair includes a lower frame portion supported so as to be capable of rocking between a standing position and a rearward tilting position with respect to a base, and an upper frame portion supported so as to be capable of rocking between a normal position and a rear end position with respect to the relevant lower frame portion. Furthermore, upper frame portion biasing that elastically biases the upper frame portion from the rear end position to the normal position is provided. This upper frame portion biasing is adapted to change an elastically biasing force to the upper frame portion corresponding to a position in the rocking movement of the lower frame portion.

1. Field of the Invention
The present invention relates to a chair whose backrest can be tilted rearward.
2. Description of the Related Art
Conventionally, there has been disclosed a chair in which a backrest surface is made up by extending an upholstery member between frame elements, which make a pair on both right and left sides. For example, there has been disclosed a chair employing a structure in which an upper portion of the backrest surface is supported by upper frame elements and a lower portion of the backrest surface is supported by lower frame elements, and these upper frame elements and lower frame elements are made independent to perform rotational operation around a horizontal axis, that is, rocking movement (for example, refer to Japanese Patent Application Laid-Open No. 2002-119375 and Japanese Patent Application Laid-Open No. 2002-119373). Furthermore, in the chair having such a constitution, by providing elastically biasing means that elastically biases the lower frame elements forward and elastically biasing means that biases the upper frame elements forward, independently of each other, the backrest surface capable of following, for example, a motion of the entire upper body of a sitter by the motion of his or her waist, and for example, a motion of only an upper portion of the upper body such as turning back and extending his or her arm laterally is realized.
In everyday life, when seated with his or her upper body standing in a chair, a sitter often turns back, extends his or her arm, twists himself or herself and so on. With respect to such motions of the sitter, in the chairs described in the Japanese Patent Application Laid-Open No. 2002-119375 and Japanese Patent Application Laid-Open No. 2002-119373, the upper frame elements follow the motions of the upper body of the sitter, particularly the upper portion with a required elastically biasing force.
However, when the sitter tilts his or her upper body rearward, the backrest is required to surely support the upper body of the sitter. That is, the elastically biasing force of such a degree that the upper frame elements of the backrest can move in accordance with the motion of the upper body of the sitter as described above does not allow the upper portion of the upper body to be stably supported. In such a chair, there may arise a defect that when the sitter inclines his or her upper body by tilting the lower frame elements of the backrest rearward, the sitter feels insecure.


An internal combustion engine includes an exhaust manifold; an intake manifold; and a turbocharger including a turbine in communication with the exhaust manifold, and a compressor in communication with the intake manifold. An electrical generator is coupled with the turbine. A motor receives electrical input power from the generator and provides mechanical output power. A combustor selectively provides additional input power to the motor.

Internal combustion (IC) engines may be of a multi-cylinder or gas turbine design. It is common for a multi-cylinder engine to include a turbocharger having a turbine which is rotatably driven by exhaust gases and a compressor providing compressed charge air to the combustion cylinders. The turbocharger uses otherwise spent exhaust gases to increase the power output from the engine.
It is known to use the exhaust gases which are discharged from the turbine of the turbocharger to rotatably drive a second turbine connected to an electrical generator. The electrical generator is selectively coupled through a controller to a motor providing additional mechanical input power to the flywheel of the multi-cylinder engine. The amount of energy remaining in the exhaust gas after passing through the turbocharger is somewhat diminished, therefore the amount of power that can be extracted with a second turbine is limited.
A gas turbine engine as mentioned above typically has a compressor stage, combustor and turbine stage. Air is pressurized in the compressor, mixed with fuel and ignited in the combustor, and then expanded in the turbine stage. A combustor is commonly used with a gas turbine engine, but is not used with multi-cylinder IC engines.
What is needed in the art is a multi-cylinder IC engine which may be operated using conventional techniques for most load conditions, but coupled to a compact auxiliary power source to provide high energy output to meet peak power demands.


A suitable the inner point is set on a vapor-liquid face of lubricating fluid formed in a composite capillary seal section that utilizes capillary force and rotational centrifugal force acted on the lubricating fluid. An inner peripheral surface of a ring-shaped portion is formed such that the minimum value R1 in radial-direction distances of the inner peripheral face of the ring-shaped portion becomes greater than a radial-direction distance R2 of the inner point (R1>R2). This can observe the inner point of the lubricating fluid from the immediately-above position, whereby the filling amount of the lubricating fluid can easily and surely be measured, thereby being capable of adjusting the filling amount of the lubricating fluid to a preset amount.

1. Field of the Invention
The present invention relates to a hydrodynamic bearing having a composite capillary seal section for preventing lubricating fluid retained in the hydrodynamic bearing from leaking to the outside of the bearing by utilizing both the capillary force and rotational centrifugal force.
2. Background Art
In a hard disk drive device or removable disk drive device, a hydrodynamic bearing has conventionally been used as a bearing of a motor for driving a recording disk, that generates dynamic pressure by lubricating fluid (for example, oil) retained in a bearing gap between a shaft and a sleeve when the motor rotates. Various types of a hydrodynamic bearing have been proposed.
This hydrodynamic bearing is required to reduce as much as possible the leakage of the oil, retained in the bearing gap, to the outside to thereby ensure the lifetime of the oil and to maintain the characteristic of the hydrodynamic bearing in a stable manner over a long period. In the hydrodynamic bearing using oil as working fluid as described above, a tapered seal gap (hereinafter referred to as a capillary seal section) is formed at the edge section of the hydrodynamic bearing, wherein oil is retained in the capillary seal section while forming a vapor-liquid surface between the oil and air in the capillary seal section.
In the capillary seal section in the hydrodynamic bearing, the size of the gap in the radial direction formed in the capillary seal section is gradually expanded as the capillary seal section is away from the hydrodynamic bearing, whereby a difference occurs in the capillary force of the oil depending upon the position where the vapor-liquid face of the oil retained in the gap in the radial direction is formed. Therefore, the capillary seal section has a function of supplying oil therefrom, in case where the amount of oil retained in the hydrodynamic bearing decreases, and a function of accommodating the increasing amount in case where the volume of the oil retained in the hydrodynamic bearing increases due to the temperature rise of the oil or the like.
A composite capillary seal section has recently been proposed, as a motor having a rotor that rotates at extremely high speed, that utilizes "capillary force" of the oil and "rotational centrifugal force acted when the motor rotates" to thereby firmly prevent the oil from flowing to the outside of the hydrodynamic bearing.
The composite capillary seal section of this type is formed in a gap in the radial direction between an inclined face 1a of a rotor 1 and an inclined face 2a of a stationary member 2, as shown in FIG. 13. The size of the gap in the radial direction between the inclined faces 1a and 2a of the rotor 1 and the stationary member 2 is gradually expanded toward an opening side (the downward side in the figure) of the composite capillary seal section 3, whereby oil 4 is retained in the composite capillary seal section 3. Further, a center axis line M of the radial-direction gap in the composite capillary seal section 3 is inclined toward the direction in which it gradually approaches the rotation axis line X, whereby upon the rotation of the rotor 1, the composite capillary seal section 3 pumps the oil 4 toward the hydrodynamic bearing (the upward side in the figure) by the rotational centrifugal force.
In the hydrodynamic bearing having the composite capillary seal section 3, the more the inclination angle of the central axis line M of the radial-direction gap in the composite capillary seal section 3 increases with respect to the rotation axis X, the more the rotational centrifugal force that pumps the oil 4 toward the hydrodynamic bearing can be increased upon the rotation of the rotor 1. Moreover, in the hydrodynamic bearing provided with the composite capillary seal section 3, the sealing function can be enhanced as the size of the radial-direction gap in the capillary seal section 3 is reduced as much as possible and the vapor-liquid face of the oil 4 is positioned in more proximate to the hydrodynamic bearing.
In this case, however, the vapor-liquid face 4a of the oil 4 retained in the composite capillary seal section 3 is shifted toward the hydrodynamic bearing, so that it is difficult to observe the vapor-liquid face 4a, i.e., the management of the filling amount of the oil 4 since the inclined face 1a of the rotor 1 interferes. As a result, in case where the filling amount of the oil 4 in the hydrodynamic bearing is small, the oil 4 may be exhausted, or in case where the filling amount is too much, the oil 4 may flow to the outside of the composite capillary seal section 3, thereby being difficult to ensure the lifetime of the hydrodynamic bearing and arising a fear of deteriorating reliability of the hydrodynamic bearing.

Hydrodynamic bearing, manufacturing method of hydrodynamic bearing, spindle motor provided with hydrodynamic bearing and disk drive device provided with this spindle motor


An image processor has a plurality of NICs (communication units), and a function of executing a plurality of jobs involving communication processes with terminals through the NICs. The job processing efficiency of the image processor is improved to the utmost extent. The types of priority jobs are preset for each of the NICs in priority job setting data, and the image processor carries out control so that a type of a job set to be given priority in the priority job setting data is processed in priority when a plurality of jobs involving communication processes through the NICs arise at overlapping timing. A job transmitter terminal transmits a job to the address of the NIC corresponding to the type of the job on the basis of the data that presets the corresponding relation between the types of jobs and communication addresses.

Generally, an image processor, such as a printer, scanner, fax machine, copier, and complex machine combining the functions of those apparatuses, has a communication unit (NIC (Network Interface Card), modem (modulator-demodulator), etc.) that communicates with an external device (computer, another image processor, etc.) via such a communication medium as network and telephone circuit. Some image processors have a plurality of communication units. In many cases, such an image processor works as so-called complex machine that has a function of executing a plurality of types of jobs involving communication processes with external devices through the communication units.
The image processor has, for example, a print function of receiving a print job from an external terminal (personal computer, etc.) through an NIC and printing out based on the print job.
The image processor has other functions, such as a scan data transmission function of transmitting image data read from a manuscript by a scanner of the image processor to another terminal through an NIC.
The image processor having a plurality of communication units is capable of executing a plurality of processes involving communication through the communication units in parallel processing.
For example, an image processor having two NICs (called a first NIC and a second NIC) is assumed. Such an image processor can execute a process involving the print function, using the first NIC, and, in parallel processing, execute a process involving the scan data transmission function (hereinafter "scan job"), using the second NIC.
Conventionally, in an image processing system including an image processor having a plurality of NICs, and terminals capable of communicating with the image processor through the NICs, the communication address (IP address, etc.) of an NIC that is a job transmission destination is preset (stored) at each terminal, so that each terminal is fixed in communication to a specific NIC. In other words, the corresponding relation between the communication units of the image processor and the terminals is fixed in the image processing system.
Japanese Laid-Open Patent Publication No. 2004-221723 discloses a technique that an image processor having a plurality of NICs sends a message to a host computer through a communication unit (NIC) that is determined to be available in sending the message to the host computer.
Regarding an image processing system including an image processor having a plurality of communication units, and terminals capable of communicating with the image processor through the communication units, however, when the corresponding relation between the communication units of the image processor and the terminals is fixed, the system poses a problem of lower job processing efficiency as described below.
When a plurality of terminals send process requests for a plurality of different types of jobs at overlapping timing to the same communication unit of the image processor, the image processor manages to execute only one of the requested jobs even if the jobs can be processed concurrently in parallel processing (e.g., print job and scan job), because the communication unit has to work on overlapping jobs. As a result, the jobs supposed to be processed concurrently in parallel processing cannot be processed concurrently, which leads to lower job processing efficiency.
On the other hand, when a plurality of terminals send process requests for the same kind of jobs at overlapping timing to different communication units of the image processor, no communication unit has to work on overlapping jobs, but the equipment other than communication units that processes the jobs (equipment composing a printing unit carrying out image formation, equipment composing a scanner unit carrying out scanning of manuscript image, etc.) has to work on overlapping jobs. The image processor, therefore, manages to execute only one of the requested jobs. In this case, a job standing by for execution may occupy a communication unit to make impossible the reception of another job that can be processed using the communication unit. As a result, the jobs supposed to be processed concurrently in parallel processing cannot be processed concurrently, which leads to lower job processing efficiency.

Image processor with communication units for processing different job types according to priority


A power supply device includes battery equipment and voltage detecting circuitry. The battery equipment includes positive-side and negative-side battery blocks that are connected to each other at a reference midpoint. The voltage detecting circuitry detects the respective voltage values of serially-connected battery modules of the battery equipment. The voltage detecting circuitry includes positive-side and negative side voltage management ICs with respect to the reference midpoint. The positive-side and negative side voltage management ICs manage the voltage conditions of the battery modules in the positive-side and negative-side battery blocks, respectively. Positive-side and negative-side voltage power lines of each of the positive-side and negative side voltage management ICs are connected to positive-side and negative-side output terminals of the battery equipment so that all the battery modules supply electric power to each of the voltage management ICs. The voltage management ICs have a common ground line.

1. Field of the Invention
The present invention relates to a power supply device such as a power supply for power tools and a backup power supply, and in particular to a power supply device that includes multiple battery modules that are serially connected to each other.
2. Description of the Related Art
Power supply devices such as a power supply for power tools and a backup power supply include multiple battery modules that are serially connected to each other in order to increase the output of the power supply devices. Thus, the power supply devices have high output. The reason is that the output of the power supply devices is proportional to the product of voltage and current of the battery modules. For example, in the case of a power tools that includes multiple battery modules that are serially connected to each other, the maximum output voltage is increased to 42 V, and in the case of a backup power supply that includes multiple batteries that are serially connected to each other, the maximum output voltage is increased to 57 V. In the case of nickel metal hydride battery power supply devices, the device includes battery modules each of which includes a plurality of nickel metal hydride batteries that are serially connected to each other. The battery modules are serially connected to each other. Thus, the nickel metal hydride battery power supply devices have high output. In the case of lithium-ion rechargeable battery power supply device, the device includes lithium-ion rechargeable batteries each of which serves as a battery module. A number of the lithium-ion rechargeable batteries as the battery modules are serially connected to each other. Thus, the lithium-ion rechargeable battery power supply devices have high output.
In power supply devices that include multiple battery modules that are serially connected to each other, the devices include a plurality of battery blocks. Battery modules are allocated to the plurality of battery blocks. The plurality of battery blocks are serially connected to each other at reference midpoints. In power supply devices that include multiple battery modules that are serially connected to each other, the devices detect the respective voltage values of the battery modules, and control charging/discharging operation. The reason is to prevent each serially-connected battery module from being overcharged and over-discharged. The electrical characteristics of batteries are likely to remarkably deteriorate if the batteries are overcharged and over-discharged. To prevent this adverse effect, a power supply device has been developed which includes circuitry that detects the respective voltage values of battery modules (see Japanese Patent Laid-Open Patent Publication No. 2003-282159). In the power supply device, battery modules that compose battery blocks are allocated to the battery modules.


A Voice over Internet Protocol (VoIP) network is described in which session state is maintained in access switches, but not signaling gateways which maintain transaction state only during pendency of a related transaction. The signaling gateway further provides transparent inter-operation between the VoIP network and non-IP networks, such as the PSTN, by means of a translator which directly translates messages between the networks.

1. Field of the Invention
The present invention relates to telecommunications system architectures, and more specifically to Voice over Internet Protocol (VoIP) architectures. Improved access methodologies and system components, including an access switch and signaling gateway, are described within the context of a novel VoIP architecture. These improved system components and methodologies manage session state more efficiently and provide a clear migration path from conventional system architectures.
2. Description of the Related Art
Although VoIP has been in existence for many years, evolving service demands are forcing a rapid evolution of VoIP technology. The pace at which new services are being integrated into existing networks continues to increase as VoIP products and services develop.
Though still evolving, packet-based telephony is becoming increasingly sophisticated. Indeed, voice protocols have developed to offer a rich set of features, scalability, and standardization than what was unavailable only a few years ago.
Critical to success of any telecommunications system is the ability to deploy value-added and high margin services. For many reason, a few of which are discussed below, VoIP and other IP-based technologies are best positioned to realize these profitable services.
Economics is perhaps the single greatest motivation for the development and deployment of VoIP. Increased competition among existing and emerging voice service vendors has brought tremendous downward pressure on the cost of voice services in the telecommunications market. This trend is likely to continue with further declines in the price of voice services.
Additionally, many existing networks support mostly data (Internet) services that are based on IP. Many services provider already own and will continue to build-out IP infrastructures. New service providers wishing to enter the voice services market will almost certainly transport voice traffic across the existing IP backbone. Building a parallel voice services network based on legacy circuit-switching equipment is simply not a cost-effective option.
Many emerging competitive local exchange carriers (CLECs) are sensitive to the cost of developing voice service networks. For many, the cost of legacy voice circuit-switching equipment is prohibitively high. Also the cost of the space, personnel, and operations capabilities required to maintain such networks is prohibitive. These carriers need a network that can be leveraged to realize other data services, such as Internet, virtual private networks, and managed network offerings. In order to provision such services and provide voice services, VoIP is an ideal solution.
Major telecommunications carriers are looking for ways to cut the cost of running and upgrading existing voice networks. These carriers want to replace and augment their existing networks with VoIP solutions for many of the reasons described above. Additionally, VoIP offers the carriers a path to circumvent existing tariff regulations. That is, carriers may use data services to transport voice calls to get around traditional (regulated) pricing structures and reduce the total cost of voice services.
In addition to cost advantages, VoIP networks offer compelling technical advantages over circuit switching networks. VoIP networks are more open to technical improvement and competition-drive improvement than circuit-switching networks which are dominated by entrenched equipment vendors. The open, standards-based architecture provided by VoIP networks allows greater interchangeability and more modularity than proprietary, monolithic, circuit-switching networks. Open standards also translate into the realization of new services that are rapidly developed and deployed, rather than waiting for a particular vendor to develop a proprietary solution.
In sum, the promise offered by VoIP technology is just beginning to be realized. As will be discussed hereafter, the contemporary state of VoIP technology and architecture is not without limitations and disadvantages. Yet, the utility offered by VoIP networks, as compared to traditional circuit-switching networks, is immense and beyond question. The issue is really one of optimally defining a VoIP architecture and providing enabling system components and access methodologies.
The advantages and preferred implementation of the present invention are best viewed and understood against the backdrop of the contemporary VoIP architectures. However, an understanding of contemporary VoIP technology requires at least some understanding of the public switched telephone network (PSTN).
There are four major tasks performed by the PSTN to connect a call. While the PSTN is capable of providing other services beyond point-to-point voice calls such services are predicated upon the following basic tasks: (1) signaling; (2) database services; (3) call set-up and tear-down; and (4) analog voice to digital data conversion.
Phones calls are inherently connection oriented. That is, a connection to the called entity must be established ahead of time before a conversation can occur. Switches, the central components in the PSTN, are responsible for creating this connection. Between the circuit switches are connections (trunk lines) that carry the voice traffic. These links vary in data communication speed from T-1 and E-1 to OC-192/STM-64, with individual channels (DS-0s) in each link type representing one voice channel. Switches are also responsible for converting the analog voice signal into a digital data format that may be transported across the network.
Signaling notifies both the network and its users of important events. Examples of signaling range from telephone ringer activation to the dialing of digits used to identify a called entity. Network elements also use signaling to create connections through the network.
The Signaling System Seven (SS7) is a standard, packet-based network that transports signaling traffic between the switches involved in the call. FIG. 1 illustrates the basic flow of a telephone call through the PSTN and SS7 network. A call begins when a user dials a destination phone number using call origination equipment 1. A local switch (not shown) typically analyses the destination number to determine if the call is a local one. If the call is local, the local switch may directly connects the call. More typically, the dialed telephone number results in a query to one or more service control points (SCP) 3. SCPs are databases that execute queries and translate the dialed telephone phone numbers into a set of circuit switching commands. SCPs also allow such common telephone features as 800 number support, 911 service, and caller ID. Signaling switch points (SSPs, not shown) are the interface between switches 4 and the SS7 network. It is here that SS7 messages are translated into the connection details required by the switches to physically connect the call origination point to the call destination point.
The SS7 control network is "out of band," that is, it is not transmitted within the same links used to carry the actual voice channels. Specialized equipment called Signal Transfer Points (STPs) 6 transport the SS7 signaling messages. As will be seen hereafter, these STPs are analogous to IP routers in that the messages are carried in data packets called message transfer parts.
The SS7 network is very expansive and has been built up over many decades of effort and expense. Indeed, the SS7 network is actually a collection of signaling networks deployed throughout the developed world. There are many technical and historical reasons why the signaling portion of the network is broken out from the rest of the system. However, the greatest value in such a design is the ability to add network intelligence and features without a dependency on the underlying circuit-switching infrastructure.
Ultimately, the destination switch signals the destination equipment 5 that a call has arrived, typically by activating a ringer. When the called party picks up the receiver and completes the circuit, a conversation may take place. Throughout the conversation, switches convert analog voice signals into digital data capable of being transported over the network.
Once the call is complete, the switches notify the SS7 network which "tears-down" the circuit connections. In contrast to the "setup" function that identifies and enables the collection of circuit connections that facilitate the call, the tear-down function disables the connections, thus freeing up the switching resources for subsequent use by the network. As one would suspect there are many more details involved in making a telephone call, but these steps describe the basic flow of events. For example, a great many supervisory messages are communicated via the SS7 network during any one completed call.
Of particular note are the concepts of "call state" and "call control." Call state is a general term referring to a great quantity of data that is maintained within the network. Call state may be stored at numerous points within the network. It includes at least data identifying the origination point of a call, the destination point for the call, and the switching control data required to connect the two points. Call state may also include data identifying particular mechanisms being used to implement and transport the call, such as analog-to-digital conversion type, echo cancellation parameters, noise elimination techniques, and silence suppression conditions, etc. Call state is maintained (stored) within the network throughout the entire pendency of the call, i.e., between call set-up and call tear-down.
Whereas call state typically defines the nature and quality of a call, call control defines "actions" associated with the call. Call control is often software code communicated in relation to a call. Execution of the code accomplishes the desired action. Actions include, as examples, commands to activate a ringer, send a dial tone, detect dialed digits, and send signaling data.
Like the PSTN, components forming a VoIP network must perform four basic functions: (1) signaling; (2) database services; (3) call connection and disconnection (bearer control); and (4) CODEC operations. FIG. 2 conceptually illustrates the VoIP network.
For purposes of this description, the IP Backbone can be viewed as one logical switch. The logical switch is, however, a distributed system. The IP backbone provides connectivity amongst a great number of distributed elements. Depending on the VoIP protocols used, this system as a whole is sometimes referred to as a softswitch architecture.
Signaling in a VoIP network is just as critical as it is in the legacy phone system. The signaling in a VoIP network activates and coordinates the various components to complete a call. Although the underlying nature of the signaling is the same, there are some technical and architectural differences.
Signaling in a VoIP network is accomplished by the exchange of IP datagram messages between network components. The format for these messages is covered by a number of standards, or protocols. Regardless of the protocol or hardware components used, the messages are critical to the function of a voice-enabled IP network and need special treatment to guarantee their delivery across the IP backbone.
Among other functions, database services are a way of locating an endpoint and translating an address communicated between two (usually heterogenous) networks linked by the VoIP backbone. For example, the PSTN uses phone numbers to identify endpoints, while a VoIP network might use a Universal Resource Locator (URL) or an IP address with port numbers to identify an endpoint. A call control database contains the necessary mappings and translations to identify endpoints. Functionally similar to call state and call control in the PSTN, "session state" in a VoIP network defines the nature of the call and controls the activities of components in the network.
The connection of a call in a VoIP network is made by two endpoints opening a communication session between each other. In the PSTN, the public (or private) switch connects logical DS-0 channels through the network to complete a call. In a VoIP implementation, this connection is a multimedia stream (audio, video, or both) transported in real time between endpoints. This connection is referred to as the bearer (or payload) channel and represents the voice or video content being delivered. When communication is complete, the IP session is ended and network resources are released.
As already noted, voice communication is analog, while data networking requires digital data. The process of converting analog voice into digital data is done by a coder-decoder (CODEC). There are many well known ways to convert analog voice into digital data. The processes used to convert, compress, and otherwise manipulate voice data in digital form are numerous and complex. Most are governed by publicly available standards.
The major components of a VoIP network are similar in functionality to the components forming a circuit-switched network. VoIP networks must perform all of the same tasks performed by the PSTN, in addition to providing a gateway between the VoIP network and the existing PSTN. Although using different technologies and approaches, some of the same component concepts that make up the PSTN are found in VoIP networks. As illustrated in FIG. 2, there are three general components forming a VoIP network: (1) media gateways 8; (2) media gateway controllers 9; and (3) the IP backbone 7.
Media gateways are responsible for call origination, call detection, and CODEC functions, including at least analog-to-digital conversion and voice packet creation. In addition, media gateways have optional features, such as data compression, echo cancellation, silence suppression, and statistics gathering. Media gateways form the interface allowing voice data to be transported across the IP network. In other words, media gateways are the source of bearer traffic. Typically, each call is a single IP session transported by a Real Time Transport Protocol (RTP) that runs over a User Datagram Protocol (UDP). Media gateways exist in several forms, ranging from a dedicated telecommunication equipment chassis to a generic PC running VoIP software.
Media gateway controllers house the signaling and control services that coordinate the media gateway functions. Media gateway controllers are responsible for call signaling coordination, phone number translations, host lookup, resources management, and signaling gateway services to the PSTN.
The IP infrastructure must ensure smooth delivery of the voice and signaling packets to the VoIP components. Due to their dissimilarities, the IP network must treat voice and data traffic differently. That is, voice and data traffic require different transport handling consideration and prioritization.
While there is correlation between VoIP and circuit-switching components, there are also many significant differences. One difference is found in the transport of voice traffic. Circuit-switching telecommunications can best be described as a Time-Division-Multiplexing (TDM) network that dedicates channels or reserves bandwidth as it is needed out of the trunk lines interconnecting an array of switches. For example, each phone call reserves a single DS-0 channel, and an associated end-to-end connection is formed to enable the call.
In contrast to the circuit-switching network, IP networks are packet-based networks based on the idea of statistical availability. Individual packets may be transported via a multiplicity of different paths through the IP network. This phenomenon often requires data packets to stop and be stored during their journey across the VoIP network. When this happens some form of queue management is required assess and handle the packets according to their priority indication. Accordingly, a packet prioritization schedule or Class of Service (CoS) definition is used to direct packets from specific VoIP applications in an appropriate manner. Such prioritization of IP network resources allows voice applications to coherently function across the IP network without being adversely affected by other data traffic on the IP backbone.
As noted above, the VoIP environment is one defined by standards. Two major standards bodies govern multimedia delivery over packet-based networks: the International Telecommunications Union (ITU) and the Internet Engineering Task Force (IETF). These two organizations are responsible for multiple protocols and equipment standards commonly used by participants in the VoIP market. A basic understanding of existing protocols is necessary to fully appreciate the present invention. These standards will be briefly described below, but a compete description may be readily obtained from numerous public sources including the organizations noted above.
Signaling System Seven SS7)
SS7 has been previously referenced in the discussion of the PSTN. FIG. 3 conceptually illustrates SS7 which is a widely used suite of telephony protocols expressly designed to establish and terminate phone calls. The SS7 signaling protocol is implemented as a packet-switched network. SS7 is both a protocol and a network designed to signal voice services. SS7 is a unified interface for the establishment of circuit-switching, translation, and billing services.
SS7 is not built on top of other protocols. Rather, as shown in FIG. 3, it is completely it own protocol suite from physical layer to application layer. For networks transporting SS7, it is important that these services be either translated or tunneled through the IP network reliably. Given the importance of SS7 signaling, it is necessary to ensure that these messages are given priority in the network. VoIP networks require access to the SS7 facilities in order to bridge calls onto the PSTN.
H.323
ITU recommendation H.323 specifies a packet-based multimedia communication system. The specification defines various signaling functions, as well as media formats related to packetized audio and video services.
H.323 standards were generally the first to classify and solve multimedia delivery issues over Local Area Networks (LAN) technologies. However, as IP networking and the Internet became prevalent, many Internet RFC standard protocols and technologies were developed and sometimes based on H.323 ideas. H.323 networks consist of media gateways and gatekeepers. Gateways serve as both H.323 termination endpoints and interfaces with non-H.323 networks, such as the PSTN. Gatekeepers function as a central unit for call admission control, bandwidth management, and call signaling. A gatekeeper and all its managed gateways form a H.323 "zone." Although the gatekeeper is not a required element in H.323, it can help H.323 networks to scale to a larger size by separating call control and management functions from the gateways.
H.323 specifications tend to be heavy handed and initially focused on LAN networking. Not surprisingly, the standard has some scalability shortcomings. One H.323 scalability issue is its dependency on TCP-based (connection-oriented) signaling. It is difficult to maintain large numbers of TCP sessions because of the greater overhead involved.
With each call initiated, a first TCP session is created using a first protocol defining a set of messages. A TCP connection is maintained throughout the duration of the call. A second session is established using another protocol. This TCP-based process allows an exchange of capabilities, master-slave determination, and the establishment and release of media streams.
The H.323 quality of service (QoS) delivery mechanism of choice is the Resources Reservation Protocol (RSVP). This protocol does not have good scaling properties due to its focus and management of individual application traffic flows. For these and other reasons, H.323 has historically been deemed ill-suited for service provider applications, and has been relegated to enterprise VoIP applications.
Real-Time Transport Protocol (RTP)
The RTP protocol provides end-to-end delivery services for data with real-time characteristics, such as interactive audio and video. Services include payload type identification, sequence numbering, time stamping, and delivery monitoring. Further, the RTP protocol provides features for real-time applications, including timing reconstruction, loss detection, content delivery, and identification of encoding schemes. Many media gateways that digitize voice applications use RTP to deliver the voice traffic. For each session participant, a particular pair of destination IP addresses define the session in order to facilitate a single RTP session for each call.
RTP is an application service built on UDP, so it is connectionless with best-effort delivery. Although RTP is connectionless, it does have a sequencing system that allows for the detection of missing packets. As part of its specification, the RTP identifies the encoding scheme used by the media gateway to digitize voice content. With different types of encoding schemes and packet creation rates, RTP packets can vary in size and interval. The combined parameters of a RTP session dictate how much bandwidth is consumed by the voice bearer traffic. RTP transporting voice traffic is the single biggest data contributor to conventional VoIP traffic.
Media Gateway Control Protocol (MGCP)
The MGCP (like its predecessor SGCP) largely defines the contemporary softswitch architecture. It is a master-slave control protocol the coordinates the action of media gateways. In effect, the MGCP divides the functional role of traditional voice switches between the media gateway and the media gateway controller.
In MGCP nomenclature, the media gateway controller is often referred to as a "call agent." The call agent manages the call-related signaling control intelligence, while the media gateway informs the call agent of service events. The call agent instructs the media gateway to setup and tear-down connections when calls are generated. In most cases, the call agent informs the media gateway to start an RTP session between two endpoints.
As one might imagine, the control and response sequence orchestrated by the MGCP between the media gateway and the call agent requires a substantial quantity of call state data to be stored at both the media gateway and call agent. The signaling exchanges between call agent and media gateway are accomplished by structured messages inside UDP packets. The call agent and media gateways typically have retransmission facilities for these messages, but the MGCP itself is stateless. Hence, a lost message is timed out by the VoIP components, as compared with a TCP delivery mechanism where the protocol attempts to retransmit a lost packet. Accordingly, MGCP messages are usually given very high priority in the VoIP network over other non-real-time data packets.
A number of routine MGCP functions executed during a call will be described in relation to FIG. 4. A call begins when a user picks up the origination phone 10 and dials a destination number. The media gateway 11 then notifies the media gateway controller (call agent) 12 that a call is incoming. The media gateway controller 12 looks up the dialed phone number and directs the media gateways to create a RTP connection (i.e., it identifies an IP address and port number). Media gateway controller 12 also informs the destination media gateway 13 of the incoming call, and the destination phone 14 rings. Thereafter, media gateways 11 and 13 open an RTP session across the IP network 15 when destination phone 14 is answered.
Session Initiation Protocol (SIP)
SIP is part of the IETF's multimedia data and control protocol framework. SIP is a powerful client-server signaling protocol used in VoIP networks. SIP handles the setup and tear down of multimedia sessions. Such sessions may include multimedia conferences, telephone calls, and multimedia content deliveries.
SIP is a text-based signaling protocol transported using either TCP (transmission control protocol) or UDP (user datagram protocol). SIP is relatively simple, efficient, and extendable, owing much of its design philosophy and architecture to the Hypertext Transfer Protocol (HTTP) and Simple Mail Transfer Protocol (SMTP). Thus, SIP uses invitations to create Session Description Protocol (SDP) messages to carry out capability exchange and setup call control channel use. Such invitations allow "participants" to agree on a set of compatible media types.
SIP supports user mobility by proxying and redirecting requests to the user's current location. Users may inform the server of their current location (an IP address or URL by sending a registration message to a registrar. Such a capability is increasingly attractive given a large and growing base of mobile users. With its mobile features, SIP implementations tend to be more discrete and SIP clients tend to be larger in number and more geographically distributed. Hence, one of the great challenges to implementing SIP services is mapping CoS delivery for the signaling and bearer traffic.


A method for ameliorating and/or preventing cancerous anemia includes administering to a subject in need thereof an agent containing 5-aminolevulinic acid (ALA) or a derivative thereof, or a salt thereof. Preferably, these ALAs contain a metal-containing compound, such as sodium ferrous citrate. The above-mentioned ALAs, ALA; various esters such as ALA methylester, ALA ethylester, ALA propylester, ALA butylester, and ALA pentylester; and hydrochlorides, phosphates, and sulfates, and the like of these ALA esters are preferred examples.

Cancerous anemia is an anemia specific to cancer patients, and is known to be an anemia largely different to generally-known iron-deficiency anemia and renal anemia. It is known that, although differing in degree, actually 80% of people suffering from cancer become cancerous anemic and the anemic symptoms worsen with cancer progression, and that anemia progresses further by anticancer drugs and radiation treatment. Cancerous anemia has become a major problem since it greatly reduces the quality of life of cancer patients.
Among that considered to be cancerous anemia, although examples of iron-deficiency anemia caused by bleeding from cancer sites are known, such is only a small portion, and the cause of most cancerous anemia is still unknown. According to Ouchi et al., although the mechanism of cancer secreting an anemia inducing factor to hemolyze red blood cells has been proposed and studied, the anemia inducing factor has still not been specified (for example, refer to Non-Patent Document 1).
In contrast, although iron-deficiency anemia in most cases is resolved by the administration of an iron supplement, iron supplements rarely show effects for cancerous anemia. Also, iron supplements are rarely used for cancerous anemia for the reason that iron promotes cancer growth.
On the other hand, it is known that the cause of renal anemia is reduction in the production of erythropoietin, which is a hematopoietic factor that promotes differentiation to red blood cells. Compounds which stimulate erythropoietin, derivatives thereof, and erythropoietin receptors have been developed and are in practical use. However, since examples in which reduction of the concentration of erythropoietin has occurred are not known for normal cancerous anemia patients, it is considered that the causes of renal anemia and cancerous anemia are completely different.
Regardless of this, many attempts have been made to use erythropoietin for the symptomatic therapy of cancerous anemia. However, although small improvement in anemia has been seen for cancerous anemia which has worsened by some anticancer drugs and radiation treatment, permission has been cancelled in the United States of America since the survival rate of the administered group was less than the placebo group, and approval has been denied in Japan. It has been reported that, according to the analysis of the experimental results, the incidence rate of thrombosis and vascular system disorder in the administered group was high, and tumor progression was promoted (for example, refer to Non-Patent Document 2).
Accordingly, although blood transfusion has been performed as symptomatic therapy in the treatment of cancerous anemia, blood transfusion has the risk of various infectious diseases and there is also the problem of iron repletion. Furthermore, since the costs are high and blood supply is limited, there is a desire for the development of an agent for ameliorating cancerous anemia and an agent for preventing cancerous anemia.

Anemia of cancer improving/prophylactic agent


Embodiments include methods, systems, and apparatus for identification, detection, and removal of cancerous cells from a patient. The apparatus includes an apparatus handle. The apparatus also includes a display including a pH measurement result. The apparatus also includes an apparatus tip including a reference electrode and a plurality of sensing surfaces, wherein each of the plurality of sensing surfaces is connected to a base of a bipolar junction transistor (BJT) device. The BJT device further includes a collector and an emitter. The apparatus also includes automation circuitry including a processing unit in communication with the apparatus tip and the display. The plurality of sensing surfaces includes a conducting material.

The present invention relates in general to surgical instruments, and more specifically, to a pH sensitive surgical tool and related operations for real-time identification and removal of cancerous cells.
When a patient tissue is diagnosed as cancerous, patients can undergo surgical procedures to have the affected tissue removed. To prevent or minimize recurrence and renewed growth of a cancerous region, it is often necessary to remove all the cancerous cells. For the surgeon, it is not possible to distinguish every cancerous cell from every non-cancerous cell with the naked eye. Thus, other modes of detection are needed to determine whether all the cancerous cells have been excised in a surgical procedure. For instance, in a conventional surgical procedure, the surgeon removes a portion of tissue that potentially contains cancerous cells and sends the portion of tissue to a pathology laboratory to determine whether it is cancerous. If it is determined to be cancerous, the surgeon removes additional tissue in a subsequent surgical excision, sometimes in the same day. In some cases, a patient and surgical team can wait in the middle of surgery for pathology results to arrive before the surgical procedure can either continue or be concluded. Multiple iterations of the surgical procedure can be needed to remove affected tissue in its entirety.
A need exists to distinguishing non-cancerous tissue from cancerous tissue during a surgical procedure to improve patient comfort and to improve patient outcomes. A need also exists to identify cancerous cells with high spatial resolution during surgery.

INTERNATIONAL BUSINESS MACHINES CORPORATION


Method and apparatuses for controlling a peer-to-peer communication involving media communication between a subscriber (A) and one or more opposite peers (B, C, D). When a policy node (300) detects that the peer-to-peer communication is activated for the subscriber, session parameters are determined (3:4) for the media communication based on predefined peer-to-peer service rules configured (3:1) at the policy node. The session parameters are then implemented (3:6) in an access gateway (304) that controls network resources for the media communication.

Various multimedia services have been developed for packet-based communication using IP (Internet Protocol), typically involving transmission of media in different formats and combinations between communication nodes such as fixed or mobile telephones, computers and servers. An architecture called "IP Multimedia Subsystem" (IMS) has also been developed by the 3rd Generation Partnership Project (3GPP) to enable such multimedia services and sessions for user terminals connected to different access networks.
Multimedia sessions can be handled e.g. by using the signalling protocol "SIP" (Session Initiation Protocol) for controlling setup of the sessions in an IMS network. A communication terminal connected to an access network may thus send a message called "SIP INVITE" to initiate a session with another communication terminal or with a server, e.g. when a multimedia application has been invoked in the terminal. The SIP INVITE message triggers different actions in the IMS network and the access network for establishing the session, including reservation of appropriate resources in the access network used.
A policy node associated to the access network is also typically used to control sessions for terminals based on various predetermined policy rules and subscription profiles, including reservation of network resources in the access network. For example, different QoS (Quality of Service) requirements can be enforced by the policy node for the above resource reservation, e.g. with respect to data bitrates and latency. The policy node is also responsible for authorising and admitting communication sessions for terminals connected to the access network, based on the predefined policy rules. In 3GPP, the policy node is often referred to as a function called PCRF (Policy and Charging Rule Function).
A media session according to a requested IP service is typically established for a user terminal by means of an application function that enables the IP service. The application function may belong to an IMS network or be implemented in a separate application server. When setting up the session, the policy node communicates various service related messages with both the application function and a gateway node in the access network in order to establish appropriate communication parameters for the session.
Sessions can also be established and handled solely by the participating parties without involving any intermediate IMS network or policy node, generally referred to as peer-to-peer (P2P) sessions, which cannot however be controlled in the manner described above. A P2P network does not have the notion of clients or servers but only equal peer nodes that simultaneously function as both "clients" and "servers" to the other nodes on the network. This model of network arrangement differs from the client-server model where communication of media is usually to and from a central server.
In a P2P session, data is transmitted in data packets between the peers and the data packets are routed through an IP network, e.g. the Internet, over various routers each having their own mechanisms and policies for handling different data flows. However, the routers are typically not concerned with what data the packets contain, i.e. the media or content being communicated, or what service is used, and there is no standardised or consistent mechanism today for controlling such peer-to-peer sessions in terms of admission and QoS related parameters.
Peer-to-peer applications on the Internet are becoming ever more common and popular, e.g. for downloading various multimedia content such as music and films. For example, a technique referred to as "Bit Torrent" is a very powerful distribution mechanism that can be used for rapid sharing of multimedia content and computer programs over the Internet without requiring a traditional centralized content server, effectively avoiding the "traffic bottleneck" associated therewith.
According to this and similar mechanisms, a tracker node is contacted to find out basically in which nodes, or "peers", a particular media object is available. After receiving a list of peers from the tracker node, different parts of the total content can be downloaded simultaneously from multiple peers in the list and these content parts are then compiled at the receiving node to form the complete content which may include one or more content files. Furthermore, so-called Distributed Hash Tables, DHT:s, are typically used in the tracker node for identifying and selecting the peers for downloading.
In FIG. 1, a typical scenario is shown for a plurality of peer-to-peer sessions between a user terminal A and multiple opposite communication nodes B, C, D, . . . over routers R in an IP network 100, e.g. for downloading different content parts from these nodes to terminal A, e.g. according to Bit Torrent. In this example, terminal A is connected to an access gateway or "edge node" EA , e.g. a GGSN (Gateway GPRS Support Node) if a mobile access network is used, while the shown opposite communication nodes B, C, D are connected to corresponding edge nodes EB, EC and ED, respectively. The transmission path for each peer-to-peer session typically runs over multiple intermediate routers R in the network 100, which may vary for different data packets during the session, which is well-known in the art. However, it is also a well-known fact that this type of mechanisms are extensively used today for illegally or illicitly spreading of content and computer programs over the Internet, i.e. without consent or control whatsoever by the content owners. Due to this lack of control, there are basically no constraints or limitations for distributing illegal or unsuitable material in this manner.
It is also a drawback with P2P sessions that no QoS can be guaranteed, e.g. in terms of transfer speed and delays, for the participating communication parties. The users are mainly dependent on the performance of underlying protocols such as sliding windows or other mechanisms implemented besides the basic P2P protocols, e.g. involving variable bitrates to create a data stream adequate for the application used. This may be a problem for real time applications where delay and jitter are critical, or when a certain download time is desirable. Another drawback associated with unrestrained P2P sessions is the amount of traffic that can be generated beyond control in the networks involved, which may result in congestion, bottlenecks, excessive processing and costs, interference and other problems.

Method and arrangement for implementing policy rules in peer-to-peer communication


The present invention discloses glasses. The glasses include a pair of temples and a frame which is integrally formed with the temples. An inner side of the junction of each of the temples and the frame is formed with a corrugated surface. The temples are provided with magnetic members in relative positions. The magnetic members engage with each other to secure the temples in place when the temples are folded. The glasses may be attracted on a metallic board or a metallic frame for convenient storage and use. The present invention discloses a glasses box which includes a box body and a box cover. The box body and the box cover are buckled together to form an accommodating space for receiving glasses. The box cover has an outer side provided with a sucker and an inner side provided with a magnet. When in use, the glasses box provides an accommodating space for receiving glasses therein. When taking out the glasses, the sucker of the box cover could be secured on a desired place. The glasses when not in use could be attracted on the magnet. This is very convenient for specific glasses which have temples made of metallic material or magnetic members.

1. Field of the Invention
The present invention relates to glasses and a glasses box.
2. Description of the Prior Art
As shown in FIG. 1, conventional glasses comprise a pair of temples 10 and a frame 20. There is no hinge between the temples 10 and the frame 20. The temples 10 are unfolded normally. As to the glasses made of resin material, for bending the temples with ease, an inner side of the junction of each temple 10 and the frame 20 is formed with a corrugated surface 30 to enhance the extensibility of the material so that the temples can be folded smoothly by applying force. The folded temples are bound together with a belt 40 so as to reduce the size of the glasses, as shown in FIG. 2. As to the glasses made of metallic material, it is not necessary to have a corrugation design. The temples can be folded directly by applying force. With a belt, the temples are bound together to reduce the size of the glasses. However, no matter what material and what shape of the glasses, it is required to use the belt for binding. This is inconvenient for storage.
In addition, a conventional glasses box is used to receive glasses. When the glasses are not in use, the glasses are folded and stored in the glasses box, preventing the glasses from being stained or broken. When needing the glasses, the glasses are opened to take out the glasses. This manner is inconvenient, particularly, to the users of reading glasses. They don't wear the glasses all the time. Sometimes, the user puts the glasses without concern, and can't find the glasses later. Probably, the glasses are stained or damaged. The glasses box doesn't provide a protection function. Even though the user puts the glasses into the glasses box, he/she may put the glasses box without concern and then can't find the glasses occasionally.
Accordingly, the inventor of the present invention has devoted himself based on his many years of practical experiences to solve this problem.


Conditional reconnection of a suspended call occurs responsive to a mobile unit (108) sending a mobile-originated reconnect (MORC) message into the network (102). Reconnection does not occur not if the mobile unit does not send a MORC message or affirmatively declines reconnection within a designated waiting period.

Wireless communication systems are well known in which mobile units can initiate or receive calls while roaming between different radio frequency (RF) coverage areas (sometimes referred to as "cells"). The mobile units communicate via RF resources with base stations distributed among the cells, which base stations are controlled by one or more mobile switching centers (MSCs). The MSCs provide control signaling for the call and connect the mobile unit to other participating endpoints, which may comprise other mobile units or wireline units.
Occasionally, mobile units can encounter service interruption(s) during a call, for example, upon entering a tunnel or reaching a fringe RF coverage area or due to a handoff error, causing the mobile unit to become dropped (at least temporarily) from the call. In such case, the MSC may attempt to page the mobile unit in attempt to find its location, reconnect the mobile unit and preserve the call. However, such network-initiated reconnects can be wasteful in terms of network resources and bandwidth, most particularly when there are multiple dropped calls and hence multiple pages flooding the network. The net effect of unsuccessful reconnection is a decreased revenue stream for the service provider.
A related problem with network-initiated reconnects is that they are performed (or not) without user input from the suspended mobile unit and the user is not informed whether a reconnect is or will be performed. Indeed, the user may not even know when RF coverage is lost because even though the mobile unit may display indicia of lost RF signal, the display might not be readily observed by the user while engaged in the call. Further, once the user believes the call is dropped, the user may reattempt the call, creating even further network traffic as the network will treat the call as a new communication session independent of the session that the network may be attempting to reconnect.
Accordingly, there is a need for a mobile-originated reconnect feature whereby a user may affirmatively request or decline a reconnect attempt rather than rely on a network-initiated reconnect that may or may not occur. Advantageously, the mobile-originated reconnect will be performed without network-initiated pages, thereby minimizing or at least reducing utilization of network resources and bandwidth; the mobile-originated reconnect will be recognized by the network as associated with a suspended (or reconnecting) call state, rather than a new call; and the mobile-originated reconnect will provide for better-informing the user of the reconnect status and RF signal status relative to the prior art. The present invention is directed to addressing these needs.

Mobile-originated reconnection of dropped wireless calls


Multilevel angle modulated information is decoded through integral use of channel state information. In a Viterbi Algorithm Trellis Code Modulation decoder (213), channel state information indicating temporal reliability of received symbols is utilized, along with the received symbol information and the Trellis Coding rules themselves, to facilitate a most probable symbol determination. The channel state information may be received signal strength indication information. The latter can be additionally processed by, for example, time delay compensation (401), normalization (403), morphological erosion (404), linearization (406), and/or through static channel detection (408).

Communication systems are known in the art. Many such systems support transmission of data from one location to another. This data will often comprise electronic symbols (such as a symbol having a particular amplitude and/or phase with respect to a baseline value) that represent particular data quantities or values. Such symbols, when properly received, can be decoded to allow reconstruction of the original message.
Radio frequency communication systems are also known, where such data is transmitted using a radio frequency channel. In a land mobile operating environment, where the sending and receiving unit are moving with respect to one another, channel degradation occurs due to such phenomena as multipath fading. Such degradation can impact symbol recognition and hence the decoding process, leading to an inaccurate interpretation of the data. Various methods are proposed in the art to protect data from channel degradation. To date, however, such solutions typically represent a compromise of one sort or the other, and hence have one or more problems associated with their adoption, including costly reception platforms to support computationally intensive recovery schemes, data throughput reduction (as necessitated, for example, by inclusion of significant error coding or signal diversity of one type or another), and varying standards of reliability as channel conditions vary.
Accordingly, a need exists for a decoding methodology that will address at least some of the above concerns.

Signal decoding method using both signal and channel state information


A semiconductor integrated circuit includes a self-testing circuit having a test circuit which is incorporated in a logic circuit to test the logic circuit. The test circuit has a test pattern generator to generate a test pattern and a compressor to compress a test result output. The logic circuit includes a plurality of scan chains including a plurality of serial connected registers and the compressor includes a through output portion. The semiconductor integrated circuit also includes a pattern counter which counts the test pattern at a test time of the logic circuit, a shift counter which counts the number of shifts in the scan chain in the logic circuit at the test time, and a failure information output circuit which is connected to the test circuit and which outputs step information of the test pattern corresponding to a failure to an integrated circuit external terminal when the failure is detected at the test time.

1. Field of the Invention
The present invention relates to a semiconductor integrated circuit (LSI), design support apparatus, and test method, particularly to a logic built-in self-test (BIST) circuit, design support apparatus, and test method, and is applied to, for example, a logic system LSI.
2. Description of the Related Art
As one of the test facilitating methods for solving the difficulty of testing a large and complicated LSI, a logic BIST is used. In the logic BIST, the generation of the test pattern given to a block to be tested, and analysis of the test result output from the block to be tested, are all automatically performed in the LSI by a logic circuit constituted in the periphery of the block to be tested.
FIG. 16 shows a connection relationships between an LSI in which is incorporated a general logic BIST circuit by a self-test using a multiple and parallel shift register sequence generator (STUMPS) system, and an external tester.
An LSI 11 includes a BIST control circuit 12, pattern counter 13, shift counter 14, logic circuit 18 including a large number of flip-flop circuits, test pattern generator 19, test result compressor 16, and the like.
The logic circuit 18 includes a plurality of scan chains 17 chain-connected to flip-flops, and is the block to be tested by the logic BIST. The control circuit 12 receives an external input signal 111 for setting a test mode from a tester 15 to set the LSI 11 to the test mode. The control circuit 12 uses output signals of the pattern counter 13 or shift counter 14 to control the block to be tested 18, test pattern generator 19, and test result compressor 16, and performs BIST control.
In the test mode, first after the logic BIST circuit is initialized, the logic BIST is executed. In this case, a test mode signal or BIST clock is supplied from the external input signal 111 via the control circuit 12, or may be directly supplied from the external input signal 111.
During execution of the logic BIST, a test pattern supplied to a scan chain 17 is automatically generated by the test pattern generator 19. Moreover, a test result output from the scan chain 17 is supplied to the test result compressor 16. In this test result compressor 16, the test result is compressed into data (signature) having a specific bit length. This test result compressor 16 supplies a signal 110 indicating a test analysis result of the logic circuit 18 to the LSI external tester 15. The tester 15 judges from the signal indicating the supplied test analysis result whether the logic circuit 18 is functioning correctly or not.
In a test in which the logic BIST circuit is used, the test pattern does not have to be prepared in a memory (not shown) of the tester 15. Therefore, the cost of the tester 15 can be reduced. Moreover, all the operations are performed in the LSI 11 in synchronization with a BIST clock signal. Therefore, when a BIST clock signal having a rate higher than the test operation frequency of the tester 15 is used, a test having a rate higher than that of the test by the tester 15 is possible. Accordingly, a product test in an actual operation can be performed.
Moreover, the test in which the logic BIST circuit is used requires only a small number of external input/output signals. Therefore, it is possible to test a plurality of blocks in parallel, and test time can greatly be reduced.
Furthermore, the test using the logic BIST circuit is not restricted by the number of scan inputs/outputs in accordance with the capability of the tester 15. Therefore, a larger number of scan chains 17 can be constructed as compared with a general scan design. When the number of the scan chains 17 is increased, the scan chain length per scan chain becomes shorter, and the test time can be reduced.
In the logic BIST circuit, in general, a logic circuit which is a test object operates at random. A random-number pattern generator is used as the test pattern generator 19. As this random-number pattern generator 19, a linear feedback shift register (LFSR) is frequently used.
FIG. 17 shows a constitution of an LFSR of five bits as one example of the test pattern generator 19 in FIG. 16.
This LFSR is constituted so that results obtained by calculating outputs 1502, 1505 of a specific register (feedback point) and output 1508 of a final-stage register with an exclusive OR gate 1501 are inputs of a top register 1515, and perform a shift operation in synchronization with the clock signal of a clock signal line 1514.
To use this LFSR as the pattern generator, an initialization operation is necessary, and all bits are set to appropriate initial values. In this initialization operation, the bit is set to the initial value held inside LSI in some case, and set to the initial value from the outside in the other case.
In the initialized LFSR, the values of registers 1515, 1503, 1504, 1506, 1507 are calculated and shifted by the exclusive OR gate 1501 in response to the clock signal. As a result, the values of the registers 1515, 1503, 1504, 1506, 1507 change at random. The random values of the registers are supplied as the random pattern to the block to be tested via output signal lines 1509, 1510, 1511, 1512, 1513.
The result of the test using the above-described test pattern is analyzed by the test result compressor 16 in FIG. 16. This test result compressor 16 is constituted, for example, of a multiple input signature register (MISR). The basic structure of this MISR is the same as that of the LFSR, but different in that the input data is taken in parallel.
FIG. 18 shows the constitution of a five-bit MISR as one example of the test result compressor 16 in FIG. 16.
In this MISR, exclusive OR gates 1610, 1609, 1608, 1607, 1606 are disposed for registers 1611, 1612, 1613, 1614, 1615 of stages. These exclusive OR gates 1610, 1609, 1608, 1607, 1606 calculate output signals of a previous-stage register (additionally, the output signal of the last-stage register 1615 for the exclusive OR gate 1610) and parallel input signals 1601, 1602, 1603, 1604, 1605. This calculation result is supplied and shifted to the respective registers 1611, 1612, 1613, 1614, 1615 in synchronization with the clock signal of a clock signal line 1616. The register 1613 receives a result obtained by calculating the parallel inputs and output signals of the previous-stage register and final-stage register 1615.
Also when this MISR is used as the test result compressor, the initialization is necessary in the same manner as in the LFSR. However, unlike the LFSR, it is also possible to initialize all the bits to "0".
When the clock signal is supplied to the initialized MISR, the parallel input data 1601, 1602, 1603, 1604, 1605 are taken in and compressed. Finally, data finally left in the register is a compression result. When the compression result is compared with an expected value (hereinafter referred to as the signature) obtained beforehand by calculation, a failure is judged.
As described above, by the test using the logic BIST circuit, presence/absence of a failure in the LSI is judged. As a result, for an LSI having the failure, the failure is sometimes analyzed.
In this case, as described above, since the information indicating the presence/absence of the failure is compressed inside the LSI in the test using the logic BIST circuit, the information necessary for analyzing the failure cannot be obtained as such. The failure analysis requires the information of a test pattern (failure pattern) for detecting the failure and scan flip-flop (failure scan flip-flop) for taking in the influence of the failure. Therefore, it is necessary to obtain this information with an operation different from a usual logic BIST operation.
When the logic BIST circuit is used to analyze the failure, there is a method of dividing the operation for each pattern. Here, one pattern indicates that a logic value captured in parallel by the scan flip-flop is serially shifted out. In general, serial shift out of a n-th pattern and serial shift in of n+1st pattern are simultaneously performed. Therefore, here, a segment of one pattern is set from when the values are captured in parallel until the logic values are serially shifted out.
In this case, the states of a test result analyzer are compared with one another for each pattern, and a pattern in which the expected value differs from the signature is identified. Therefore, the test pattern for detecting the failure is obtained.
Next, to know the position of the failure scan flip-flop, the scan chain 17 of the block to be tested 18 shown in FIG. 16 is switched to a scan test mode.
FIG. 19 shows a constitution example of the scan test mode of the logic BIST circuit shown in FIG. 16.
An LSI to be tested 1701 is switched to the scan test mode by a control signal 1703 from a tester 1710, and a scan chain 1706 is cut off from a test pattern generator 1709 and test result compressor 1705. This scan chain 1706 is connected to one or several scan chains.
A scan input terminal 1708 and scan output terminal 1704 are connected to a scan channel of the tester 1710. In this scan test mode, the test pattern obtained beforehand is scanned in or out from the outside to specify a failure scan flip-flop.
However, when the logic BIST circuit is used to analyze the failure as described above, there are several problems as follows.
That is, an execution result of the tester has to be analyzed in order to obtain the test pattern. In addition to this analysis time, a test time lengthens, because the mode is switched to the scan test mode as shown in FIG. 19 to carry out the test a plurality of times.
Moreover, it is necessary to separately prepare the scan test pattern, and as a problem, a design time required until the scan becomes possible lengthens.
Furthermore, the mode is switched to the scan test mode to carry out the test. Therefore, the failure which can be detected only by a logic BIST operation cannot be analyzed. This is an important problem, when an actual operation rate test is carried out by the logic BIST operation. Therefore, there has been a semiconductor integrated circuit in which during the test performed using the logic BIST circuit, it is possible to output number information of the pattern in which the failure is detected, and further the position information of the scan flip-flop as circumstances demand, and a failure analysis operation can easily be performed outside. Furthermore, to design the logic BIST circuit, there has been a demand for development of a design support apparatus of a semiconductor integrated circuit, which can easily prepare a file. The file includes a correspondence between failure count and failure pattern, in which conversion of the failure count to the failure pattern is possible.

Semiconductor integrated circuit, design support apparatus, and test method