Abstract: This invention provides the ability to significantly lower the power consumed by a group of Ethernet links when organized in a Link Aggregation Group. When the server or switch senses low bandwidth utilization across multiple links in the group, it will negotiate the transition of un-necessary links to a lower power state. When the bandwidth requirements increases, the algorithm will quickly re-establish links and distribute the Ethernet traffic across the multiple links when necessary.

Abstract: The present invention discloses one or more network components within a telecommunication network wherein said network is a serial bitstream network and the network components are comprising one or more inverse multiplexers, wherein said multiplexer(s) is/are TDM bonding multiplexers and said multiplexer(s) utilize(s) spare bits and/or FAS bits in a protocol for control of specific functions of the inverse multiplexer(s). The invention further includes a method for inverse multiplexing of one or more serial bitstreams within a telecommunication network wherein said method uses spare bit and/or frame alignment signals from a protocol to: measure differential delay between transmission lines at the receiving end, and/or secure timeslot integrity by compensation for individual link delay, and/or secure correct transmitter and receiver timeslot sequence by link identification, and/or monitor channel availability by block error checking.

Abstract: In one embodiment, a message generation scheme is disclosed that is operable with respect to user data information relating to the signaling of an IMS message over a CS network. A number of User Data Items (UDIs) may be configured at one of the entities, e.g., a UE device or a network node, wherein each UDI is operable to contain a plurality of Information Element Identifiers (IEIs). For a select IEI, an Information Element Data Length (IEDL) structure is provided that is operable to indicate a length of data associated with the select IEI, wherein the data comprises at least a portion of the user data information. One or more alternative signaling (AS) messages are generated for transmission to a second entity, wherein the AS messages include one or more of the UDIs.

Abstract: Methods and apparatus for communication fabric bandwidth management are disclosed. An example method includes receiving data at a first network entity, where the data being received from a second network entity. The example method further includes, at the first network entity, queuing the received data in a data queue associated with the second network entity. The example method still further includes determining that an amount of queued data in the data queue associated with the second network entity exceeds a first threshold. In response to the first threshold being exceeded, a first control message is communicated from the first network entity to the second network entity. In the example method, in response to the first control message, a data rate at which the second network entity sends data to the first network entity is reduced.

Abstract: The presented invention provides methods and apparatus to generate a efficient medium dedication schedule which give wireless medium the ability to serve real-time and non real-time application concurrently and still be able to maintain the QoS as requested by real-time application. Furthermore, it also presents methods for implementing buffer in an efficient matter in order to complement the advantage introduced by burst transmission and burst acknowledgement.

Abstract: A vestigial sideband (VSB) modulation transmission system and a method for encoding an input signal in the system are disclosed. According to the present invention, the VSB transmission system includes a convolutional encoder for encoding an input signal, a trellis-coded modulation (TCM) encoder for encoding the convolutionally encoded signal, and a signal mapper mapping the trellis-coded signal to generate a corresponding output signal. Different types of the convolutional encoders are explored, and the experimental results showing the performances of the VSB systems incorporating each type of encoders reveals that a reliable data transmission can be achieved even at a lower input signal to noise ratio when a convolutional encoder is used as an error-correcting encoder in a VSB system.

Abstract: After a predetermined period of time has lapsed after transmission of a voice data packet representing a first voice section a second voice data packet representing the same voice section is transmitted. On the receiving end, a value for the transmission quality is determined for the first and for the second voice data packet and the one of the two voice data packets is processed that has the better transmission quality.

Abstract: A spectral reuse transceiver-based communication system conducts communications between a master site and a plurality of remote sites using a selected portion of a communication bandwidth containing a plurality of sub-bandwidth channels. Each remote site transceiver monitors the communication bandwidth for activity on the sub-bandwidth channels, and informs a master site transceiver which sub-bandwidth communication channels are absent communication activity and therefore constitute clear channels. The master site transceiver compiles an aggregate list of clear channels from all the remote sites and then broadcasts the aggregate list to the remote sites. The master site and a remote site then conduct communications therebetween by frequency-hopping and/or orthogonal frequency multiplexing among the clear channels using an a priori known PN sequence.

Abstract: A method for minimizing bandwidth consumption for communicating over an interface is provided that includes defining a one-frame event that comprises a specified operation type and defining an n-frame event that comprises the specified operation type. A single frame is assembled for the one-frame event. The single frame comprises an event type identifier that is operable to identify the single frame as corresponding to the one-frame event. The single frame is sent over the interface.

Abstract: A technique for compressing header information to produce a compressed header portion of a data packet to be transmitted across a communication channel includes generating a checksum from the header information, and providing the checksum as the compressed header portion.

Abstract: A compressor/decompressor (Codec) selecting apparatus includes a performance analyzer, a training server, a storage unit, and a selecting unit. The performance analyzer analyzes the performance of the current internet communication system, and outputs environmental parameters accordingly. The training server obtains learning data corresponding to a hyperplane according to training samples. The storage unit stores the learning data and the related parameters. The selecting unit does a functional analysis according to the environmental parameters and the learning data in order to select the Codec suitable for the current internet communication system.

Abstract: Methods and apparatus are provided for allocating bandwidth among members of two statistical multiplexing groups having overlapping members. At least two groups of such services having overlapping members are defined. The members of a first one of the groups consume a first bandwidth and the members of a second one of the groups consume a second bandwidth. A total bandwidth of all the members of the groups together is restricted to be no more than a total available bandwidth. In order to accomplish this, the members of the first and second groups are statistically multiplexed based on bandwidth requirements of members common to both said first and second groups.

Abstract: For steady state voice data packet transmission between a mobile station and a packet data service node a new compressed TCP/IP header (160) concatenated with a compressed RTP/UDP header (4) is sent. This concatenated header is seven bytes in length instead of the typical 40 byte long RTP/UDP/IP header. A new TCP header arrangement (30) transmits a special access code (161) to a Van Jacobson TCP/IP header compression/decompression arrangement (20). This allows the voice data packet to be transmitted to the receiving end without the other 33 bytes of header information. The PDSN regenerates the IP header and the receiving end then regenerates the RTP/UDP header (205) while it discards the new TCP header arrangement (30).

Abstract: Architecture for providing a compressed replacement media file to a mobile device is provided. The method comprises receiving a first media file of a first format. The first media file is then compressed to generate a second media file of a second format. A request may be received from a mobile device for the first media file and the second media file is transferred to the mobile device via the Internet. Content may be added to the first media file such that the second media file comprises both the added content and the first media file. An indication that the second media file has been played on the f media device may be received.

Abstract: The present invention is directed towards compressing and decompressing data transmitted between an end terminal (12) and a network server (20) controlling compression in communication sessions involving the end terminal. States related to the end terminal that are created and used for compressing and decompressing data transmitted between the end terminal and the network server when the end terminal is first registered to the first network server are saved as the end terminal deregisters from the first network server. When the end terminal again registers to the network server these saved states are directly used. In this way it is possible to start compressing data using the saved states immediately after a following registering for providing a more effective dynamic compression.

Abstract: Contiguous data packets of a single communication session between endpoints are gathered for compression, and at least the payloads are compressed collectively via a single compressible buffer. The original headers, which may or may not be compressed, and the compressed payloads are transmitted from a sending packet compression device to a receiving packet compression device that performs decompression to the compressed payload and may also decompress headers if they have been compressed at the sending end. The original contiguous data packets including headers and payload may be reconstructed at the receiving packet compression device by matching the headers to the corresponding payloads. The reconstructed data packets are injected back into the single communication session, and the presence of the original headers in the reconstructed data packets may thereby maintain the endpoint-to-endpoint authentication protocols.

Abstract: This application is related to a method and apparatus for optimizing the transmission of PLMN-IDs in a wireless network. This is accomplished by reducing the amount of bandwidth required to transmit a given set of PLMN-IDs.

Abstract: A new system and method of available bandwidth estimation applies even where the narrow link and tight link in a network path are in different locations. In embodiments of the invention, a unique packet probe series structure and processing is employed to estimate available bandwidth. In an embodiment of the invention, the spacing between probe packets is adjusted at the source to account for dilation caused by links leading to the tight link, so that the spacing is appropriate when the probes arrive at the tight link. Moreover, the multi-packet probe comprises a large packet followed by two much smaller packets. The large packet is then dropped once it has traversed the tight link. The two small packets, which are impacted little by subsequent narrow links, preserve the spacing set by the tight link, encoding the delay induced by the tight link, all the way to the destination.

Abstract: A method for transmitting digital voice data packets between voice data inputs of a plurality of channels (A B, C, D) and a transmitter comprises the following steps: compressing the voice data packets to compressed data packets in a vocoder assigned to every channel, feeding the compressed data packets of every channel to a control device, unsolicited, unsynchronized transmission of the compressed data packets on a data bus common to all channels from the control device to a processor assigned to the transmitter, combining a defined number of compressed data packets of a certain channel which have been transmitted last on the data bus to a transmission block by means of a processor, a last compressed data packet being fed to the transmission block only when the transmission of the transmission block has already started.

Abstract: A bandwidth allocation system configured for in-service upgrading of bandwidth allocation functionality comprises dynamic bandwidth allocation software, a first processor, a second processor, first memory and second memory. The bandwidth allocation software includes an image representing firmware configured for determining bandwidth allocation requirements. A first processor interprets the firmware for facilitating determination of bandwidth allocation settings for a plurality of optical network termination units of a PON system. The second processor interprets the software for managing information upon which the bandwidth allocation settings are dependent.

Abstract: A method and computer program product for providing RTP suppression across a DOCSIS network. An index number and a set of rules are sent to a receiver. The index number indicates the type of header suppression technique (i.e., RTP header suppression) to be performed, and the set of rules define how to recreate the RTP packets on the receiving end. At least one complete RTP packet is transmitted upstream for enabling a receiver to learn the RTP header. Subsequent RTP packets are transmitted upstream for reconstruction at the receiving end. The subsequent RTP packets are comprised of delta values representing fields that dynamically change from packet to packet in an RTP header.

Abstract: Disclosed is a device for filtering out a null packet in a data stream for example an MPEG-2 TS (Transport Stream) to avoid bandwidth waste when it is transmitted to a backbone network. The device includes a plurality of receiving interfaces, a filter unit, and a controller. The receiving interfaces receive parallel data, such as DVB-ASI (Digital Video Broadcasting-Asynchronous Serial Interface) inputs as channel-by-channel MPEG-2 data. The filter unit counts the respective numbers of data packets and non-data packets of the data input from the receiving interfaces and filters a null packet in the data. A network transmission unit transmits the filtered data to a backbone network. The controller calculates a bit rate based on both of the count numbers received from the filter unit.

Abstract: A object of the present invention is to decrease or avoid the distortion of an MC-CDMA signal transmitted over a reverse link channel for a given amplifier efficiency. The present invention concerns a code allocation method for a mobile telecommunication system. According to the invention, for each of a plurality of available spreading codes or available combinations of spreading codes a value of a first variable (PAPRk) characteristic of the dynamic range of a modulated signal (Sk) is determined, and for each of a plurality of users a value of a second variable (?k) characteristic of the propagation loss incurred over the transmission channel of the user is determined. A spreading code or combination of spreading codes producing a low dynamic range is allocated to said user if the propagation loss over its transmission channel is high.

Abstract: Aspects of a method and system for reducing transceiver power via a variable number of channels are provided. In this regard, utilization and/or availability of network and/or device resources may be determined and a configuration of channels utilized for transmitting data may be determined based on the determined utilization and/or availability of resources. Accordingly, the number of channels over which data is communicated may be altered based on determined thresholds for the utilization and/or availability of resources. Moreover, the configuration of channels utilized for communicating data may be dynamically altered by monitoring changes to the utilization and/or availability of resources. For example, the number of channels may be altered based on available bandwidth on one or more channels, based on capacity and/or available space of one or more buffers, and/or based on available power or desired power consumption of a transmitter.

Abstract: A method is described for transmitting a byte stream (1) from a transmitter unit to a receiver unit in a network protocol, with the following steps: Removal of at least one data area (4a,4b) of the byte stream (1) from the byte stream, Transmission to the receiving unit of information relating to which data area (4a,4b) or which data areas (4a,4b) has or have been removed from the byte stream (1) and/or relating to the way in which the shortened byte stream is to be further processed, Transmission of the byte stream (5) shortened by at least one data area (4a, 4b) to the receiver unit, which, from the information and the shortened byte stream, determines and further processes the at least one data area removed by the transmitter unit.

Abstract: An adaptive compression mechanism that dynamically selects compression algorithms applied to network application traffic to improve performance. One implementation includes an arbitration scheme that reduces the impact on computing resources required to analyze different compression algorithms for different network applications. The adaptive compression functionality of the present invention can be integrated into network application traffic management or acceleration systems.

Abstract: A controller (103a, 103b) provides dynamic capacity, for use in connection with a communication network (101). A processor of the controller (103a, 103b) that can be utilized in the communication network can be configured to facilitate determining whether a current load level is below a predetermined low usage, and responsive thereto, adjusts a bearer configuration to allow higher throughput per user. Further, the processor (103a, 103b) can be configured to facilitate determining whether the current load level exceeds a predetermined high usage, and responsive thereto, adjusting the bearer configuration to allow lower throughput per user.

Abstract: A method of route distribution for implementing multi-service optimization in Synchronous Digital Hierarchy Transmit Network comprises the following steps: A. dividing the SDH transmit network into subnets in term of rings to form a plurality of ring subnets, calculating initial routes of all services demand in the SDH transmit network; B. judging whether there are link resource overload between subnets and inside subnets respectively, if yes, recalculating the route; C. judging whether the load balance index is satisfied by ring subnet, if yes, performing route adjustment; D. after dividing the subnets into edge subnets and core subnets, judging whether the aggregative indicator of each link in the core subnets satisfies convergence condition, if no, checking whether the route calculation times exceeds the threshold, if yes, ending; otherwise, recalculating the corresponding routes.

Abstract: A method of transmitting data packets, in particular a real-time or near-real-time data over an internet protocol (IP) network using a real-time protocol (RTP) for media data packets and a real-time control protocol (RTCP) for control data packets. Each protocol is allocated a fraction of the available transmission bandwidth. The method comprises the steps of determining the needed RTCP bandwidth based on known link parameters for an individual media session, transmitting control data packets using the needed RTCP bandwidth and transmitting media data packets using the remainder of the available bandwidth.

Abstract: A method for shifting the position of a symbol to reduce transmission overhead is described. A frame with a plurality of symbols is provided. A determination is made whether the frame includes an indication of an event. At least one symbol of the plurality of symbols is shifted based on the determination. A counter is set to an initial value. The frame with the at least one shifted symbol is transmitted.

Abstract: The present invention provides a TCP/IP header compression format as a label in package transmitting. The header format is used under the acknowledgement mode in wireless network to reduce the header transmission amount between the base station and the mobile station, comprising: a connecting sequence number field for storing a connecting sequence number to indicate the required TCP/IP header information related base stations; a buffer length field for storing a length value of the buffer region to indicate the acceptable package number of the buffer region in receiver; and a plurality of control flag fields for providing certain necessary information for transmission control.

Abstract: A home wireless router establishes a Wireless Local Area Network (WLAN) and services a plurality of wireless terminals within the WLAN service area including at least one Voice over Internet Protocol (VoIP) wireless terminal. The home wireless router receives a request from the VoIP wireless terminal to establish a VoIP conference call with at least two other VoIP wireless terminals. The home wireless router interacts with the VoIP service accumulator to determine at least one other home wireless router required to service the VoIP conference call, queries the at least one other home wireless router to determine its ability to service the VoIP conference call for at least one serviced VoIP wireless terminal, and receives a response from the at least one other home wireless router. The home wireless router, the at least one other home wireless router, and the VoIP service accumulator then set up and service the VoIP conference call.

Abstract: Disclosed are an apparatus and method for providing packet data service from a radio network controller (RNC) to user equipment (UE) in a mobile communication system. A radio bearer (RB) of a predetermined channel type is configured to compress and de-compress headers of packets passed between the RNC and the UE. The packet data service is provided from the RNC to the UE via the RB. When the channel type is changed, configuration information associated with an RB of the changed channel type is determined. Header compression (HC) context used in the RB of a previous channel type is acquired according to the configuration information and the RB of the changed channel type including the acquired HC context is configured. The packet data service is continuously provided via the RB of the changed channel type. Therefore, the invention reduces waste of radio resources that is due to a channel type change and delay that is due to system reset.

Abstract: A time-shared network comprising edge nodes and optical core nodes may be dynamically divided into several embedded networks, each of which covering selected edge nodes. At least one of the edge nodes may host an embedded-network controller operable to form multiple-source flow-rate allocation requests each of the requests specifying flow-rate allocations to a plurality of paths from several source nodes to several sink nodes. A core node may also host an embedded-network controller or several embedded-network controllers. The time-shared network may use both time-division multiplexing and burst switching.

Abstract: In a system including a parent station and a plurality of child stations connected only to the parent station, which is unable to detect a response among the plurality of child stations, the parent station includes a transmission permission sequential allocation mechanism for giving transmission permission sequentially and a first timer for monitoring a first period which is a response detection time period. The child station includes a transmission permission detection mechanism for detecting that the identifier sent by the parent station is directed to the own station. Each of the child stations transmits data in a band period when the child station is permitted to communicate. When the parent station does not receive a response from the child station permitted to communicate within the first period, the parent station permits a next child station to communicate.

Abstract: A data processing apparatus and method and a computer readable storage medium storing data processing programs realizing the data processing method, capable of preventing transmission of unnecessary redundant data and efficiently utilizing a transmission path. In the data processing method, data is input, the data is compression-encoded, the encoded data is packetized into a first data train on a basis of a first data length, the generated first data train is packetized into a second data train on a basis of a second data length, and the first data length is controlled in accordance with the second data length.

Abstract: A wireless LAN comprises an access point with a data communicator for data communicated over different channels, each using a respective wireless technology, and at least one mobile communications device with a data communicator for data communicated over the channels and using the wireless technologies. A first of the channels uses a wireless technology operating at a first frequency bandwidth, and a second of the channels uses a different wireless technology operating at a second, non-overlapping frequency bandwidth. The wireless technology used for the downlink channel operates at a higher data rate than the wireless technology used for the uplink channel. The controller controls data communications over the downlink channel and the uplink channel to maximise the downlink data communication QoS.

Abstract: A method is disclosed for transmitting and receiving packetized data having multiple protocol headers in a packet data communication system. An operating state of a network is determined, and a period for transmitting a full packet is decided according to a result of the determination. Full packets having uncompressed headers are transmitted during periodic transmission times according to the decided full-packet transmission period, and compressed packets are transmitted in other transmission times. If the operating state of the network is decided to be a congested state according to an average packet retransmission ratio produced in a period for measuring the operating state of the network, the full-packet transmission period is set to “1”. A value of the full-packet transmission period in an unloaded state is twice a value of the full-packet transmission period in a normal state.

Abstract: Systems and/or methods for relaying messages between nodes in a network (e.g. a wireless network) are provided. In accordance with certain exemplary embodiments, substantially simultaneous communications between nodes may be accomplished. At least one secondary message may be piggybacked onto a primary message. Messages in the network may be queued and sent from a transmitter to at least one receiver based at least in part on the signal-to-noise ratio(s) of the receiver nodes with respect to the transmitter. The queuing of the messages may be performed in dependence on a link set that indicates which messages are capable of piggyback communications. Thus, it may be possible to realize a network that reduces collision problems, reduces delays in communications, and/or increases throughput.

Abstract: A header compression in a broadband wireless communication system is provided. A transmitter includes a compressor for compressing a header of a packet; a processor for appending a compression channel IDentifier (ID) to the packet; and a controller for identifying a Service Flow (SF) corresponding to the compression channel ID and processing the packet according to the SF. Accordingly, by using the ID of the compression channel in the header compression, the SF can be identified even in the header compression.

Abstract: Systems and/or methods for relaying messages between nodes in a network (e.g. a wireless network) are provided. In accordance with certain exemplary embodiments, substantially simultaneous communications between nodes may be accomplished. At least one secondary turbo-coded message may be piggybacked onto a turbo-coded primary message. Messages in the network may be queued and sent from a transmitter to at least one receiver based at least in part on the signal-to-noise ratio(s) of the receiver nodes with respect to the transmitter. Thus, it may be possible to realize a network that reduces collision problems, reduces delays in communications, and/or increases throughput.

Abstract: A cable modem system and method is provided for supporting data transfer protocols that extend beyond standard protocols used in conventional data over cable systems. A cable modem in accordance with the invention determines, during registration, whether a CMTS with which it communicates is capable of supporting an extended protocol, and if it is, transfers data to the CMTS in accordance with the extended protocol. Additionally, a CMTS in accordance with the invention is notified during registration whether or not a cable modem supports an extended protocol and stores this information. When a request for transmission opportunity is subsequently received from the cable modem, the CMTS accesses the stored information to determine if the cable modem supports the extended protocol. If the cable modem supports the extended protocol, the CMTS processes data received from the cable modem during the transmission opportunity in accordance with that protocol.

Abstract: A cable modem system and method is provided for the transfer and processing of data in accordance with specialized data transfer protocols while utilizing conventional cable modem termination system (CMTS) equipment. A cable modem system in accordance with the invention includes a cable modem that provides for the modification of data packets in accordance with a proprietary protocol and the addressing of the modified data packets to a headend server. The headend server is adapted to reconstruct the data packets for transmission to other local cable modems or to a packet-switched network. In additional embodiments, the CMTS itself rather than a headend server operates to reconstruct the data packets.

Abstract: In wireless networks, data objects are typically transferred from a data distributor to a data recipient by reliable point-to-point communication between the distributor and the recipient. However, the time to transfer data by point-to-point communication can become excessive if the number of recipients is large. Efficient and reliable data transfer is provided by a system incorporating broadcast and point-to-point communication topologies. The data is initially broadcast to all recipients. Data broadcasting is followed by polling the recipients. If the data missing after a broadcast can be transferred in less time by point-to-point communication then that topology is chosen to complete the transfer. If not, the missing data is broadcast again to all recipients. The efficiency of the broadcast phase can be improved by designating a representative to acknowledge successful data transfer on behalf of all recipients.

Abstract: A system processes data units in a network. The system receives a data unit that includes a group of headers and suppresses one or more of the headers to form a reduced data unit. The system suppresses one or more other headers of the reduced data unit to form a further reduced data unit and transmits the further reduced data unit to one or more destination devices using the program identifier (PID) field in the MPEG header as an index to suppressed headers.

Abstract: An RTP payload header (408) is compressed by utilizing a codebook (514) that holds one or more indexable payload header variations and sending it to a receiving device (504). An RTP packet (400) that includes the payload header (408) with information pertaining to the packet is generated by a sending device (502) and the codebook is searched for the payload header information. If the information is found, the payload header (408) is replaced with a short index in the codebook (514). At the receiving device (504) the index is used to retrieve a payload header variation that corresponds to the index and the variation is placed back into the payload header (408) to uncompress the header.

Abstract: A distributed buffering system includes at least one input buffer, at least one serializing module, a at least one deserializing module, at least one output buffer, and a programmable logic device. The input buffer is operably coupled to store at least one data block of incoming data. The serializing module serializes the data block as it is retrieved from the input buffer to produce a serial stream of data. The programmable logic device receives the serial stream of data and distributes it to one or more of the at least one deserializing modules. The at least one deserializing module converts the serial stream back into the data block. The recaptured data block is then provided to the corresponding output buffer, which stores the recaptured data.

Abstract: Techniques for reducing overall information loss and processing costs due to data translation in a network populated with communicating data sources and sinks of multiple data representations is described. By employing an “optimization” technique for negotiating “least cost” data representations to be communicated between data source(s) and sink(s) where the “least cost” can be relative to, for example, fidelity requirements and processing costs of the system or the user of the system.

Abstract: A system 10 for distributing content includes a content provider 12 that provides content through various devices including a satellite 26 or a terrestrial-based system 38. The content provider 12 provides content to a content distribution system 70 within a vehicle 60. The method associated with the system includes generating a request for a list of content from a terminal, providing a list of content to a terminal, selecting a selection from the list of content, transcrypting or transcoding the content, streaming the content corresponding to the selection to the terminal and displaying the content on the terminal.

Abstract: A system that packetizes telephone calls using loss-less codecs for transmission over a communication link. The system includes a call admission control mechanism that admits calls for transmission over the communication link. An adaptive system which monitors bandwidth usage and which dynamically provides a feedback loop to the CAC system. The bandwidth allocation for transmission of the calls and the admission of calls is based upon actual bandwidth usage conditions in the system.