Showing papers on "Transmission delay published in 1995"

Method for hashing in a packet network switching system

Abstract: The port in a packet network switching system that a packet should be associated with is determined by retrieving packet address information for a packet that is to be transmitted. A predetermined number of bits from the packet address information is selected to use a hash key, which is used to compute a table address. The contents of the table at that address are compared with the packet address information. If it matches, the packet is transmitted over the port associated with that particular destination address. If it does not match, the table address is incremented by one, and the contents of the new table location identified by the incremented address are compared with the packet address information. A high speed digital video network apparatus which utilizes the hashing function is implemented on a single integrated circuit chip, and includes a network protocol processing system interconnection, compression/decompression circuits, and encoder/decoder circuits.

A system for transmitting packet data in radio telephone TDMA systems

Abstract: The invention relates to a method for transmitting packet data in a cellular system. The number of time slots in a TDMA frame dedicated for packet transmission varies according to transmission needs and each logical channel consisting of corresponding time slots in consecutive TDMA frames is independent of the other logical channels. A data packet is encoded in an information channel frame consisting of N-1 information bursts, and between the frames there can be an acknowledge/retransmission request burst (ARQ) reporting that a received frame was error-free or requesting retransmission. Thus, the information channel consists of repeated sequences of N bursts. Also disclosed are the structure of a packet paging burst (PP), packet random access burst (PRA), packet access grant burst (PAG), acknowledge/retransmission request burst (ARQ) as well as the use of the bursts in starting and maintaining the transmission.

Distributed processing ethernet switch with adaptive cut-through switching

Abstract: A packet switching system includes at least two network cards each receiving data packets via a plurality of associated ports, a system card, and an interconnect for connecting the system card to the network cards. Each one of the network cards comprises a plurality of port controllers for sending and receiving packets to and from a corresponding port and a packet processor for buffering packets received by the port controllers. The packet processor then sends destination addresses to the system card via the interconnect and receives forwarding information from the system card. The processor then forwards the packet in response to the forwarding information. The processor begins forwarding the packet in response to the forwarding information before the packet has been entirely received and checks the integrity of the packet by reference to check sum information contained in the packet as in cut-through switching. Future packets from the source port have their validity checked prior to forwarding in response to receiving an invalid packet from the source port as in store and forward switching.

System for transmitting packet data in digital cellular time division multiple access (TDMA) air interface

Abstract: The invention relates to a system for transmitting packet data in the air interface of a digital cellular system based on TDMA, Time Division Multiple Access. The mobile terminated logical channels comprise information channels designated for transmitting information and control channels, which can be a fast paging (FP) channel and an acknowledgement (A) channel. The mobile originated logical channels comprise information channels designated for transmitting information and a reservation (R) channel, whereon the mobile station requests the system to reserve a connection for transmitting packet data. According to the invention, for the TDMA frames there is allocated a variable number of time slots for packet transmission, taking into account the symmetricity/asymmetricity of the packet transmission, as well as the total packet transmission demand of the cell. For fast paging (FP), acknowledgement (A) and reservation (R), there can be employed any of the time slots in the frame allocated for packet transmission. It is advantageous that in each time slot, the subscriber's data is subjected to the same interleaving and forward error coding algorithm, so that the respective time slots of consecutive frames form independent logical sub-channels, which are then reserved for one subscriber according to the needs, and to which the subscriber's data is multiplexed at the beginning of the transmission, and wherefrom it is again demultiplexed after the transmission.

Synchronization system for networked multimedia streams

Abstract: A network data stream synchronization method and system are disclosed. Operating delay and loss parameters are accepted and a data stream buffer is initialized. The network delay experienced by a data packet is determined and a buffer delay is imposed to provide a fixed end-to-end delay. Packets arriving too late to be played within the fixed end-to-end delay are discarded. Network delay models may be updated and clock drifts may be detected and compensated for using historical buffer data.

Method for ensuring synchronization of MPEG-1 data carried in an MPEG-2 transport stream

Abstract: A method for ensuring synchronization of MPEG-1 data is carried in an MPEG-2 transport stream for decoding by an MPEG-1 decoder. The method includes searching the MPEG-2 transport stream for a packet start code prefix indicating the start of a PES packet and storing the data associated with the PES packet into a buffer. If the PES packet is a video PES packet, the packet header is examined to determine if a PTS is present. If a PTS is present, the video data in the PES packet are examined during storage to the buffer to locate the first occurrence of a picture start code in the data. Once the first occurrence of a picture start code is located, a flag is set indicating the correspondence between the PTS and the identified picture start code. The video data remaining in the video PES packet is then stored to the buffer without further examination. The PES packet data are stored without examination if a PTS is not present. In either instance, the stored data are transmitted from the buffer to the video decoder in known lengths. If the PES packet is an audio PES packet, a current value of a system time clock is subtracted from a value of a first PTS contained in an audio PES packet to calculate an audio delay. The audio delay is then adjusted to account for audio decoding, audio transmission, and video decoding times. Audio data stored in the buffer are not enabled for delivery to an audio decoder until after the audio delay has elapsed.

Single chip network router

Abstract: A single chip router for a multiplex communication network comprises a packet memory for storing data packets, a Reduced Instruction Set Computer (RISC) processor for converting the packets between a Local Area Network (LAN) protocol and a Wide Area Network (WAN) protocol, a LAN interface and a WAN interface. A Direct Memory Access (DMA) controller transfers packets transferring packets between the packet memory and the LAN and WAN interfaces. A packet attribute memory stores attributes of the data packets, and an attribute processor performs a non-linear hashing algorithm on an address of a packet being processed for accessing a corresponding attribute of said packet in the packet attribute memory. An address window filter identifies the address of a packet being processed by examining only a predetermined portion of said address, and can comprise a dynamic window filter or a static window filter.

Analysis of Audio Packet Loss in the Internet

Abstract: We consider the problem of distributing audio data over networks such as the Internet that do not provide support for real-time applications. Experiments with such networks indicate that audio quality is mediocre in large part because of excessive audio packet losses. In this paper, we show using measurements over the Internet as well as analytic modeling that the number of consecutively lost audio packets is small unless the network load is very high. This indicates that open loop error control mechanisms based on forward error correction would be adequate to reconstruct most lost audio packets.

Delay guarantee of virtual clock server

Abstract: In a packet switching network, each communication channel is statistically shared among many traffic flows that belong to different end-to-end sessions. We present and prove a delay guarantee for the virtual clock service discipline (inspired by time division multiplexing). The guarantee has several desirable properties, including the following firewall property: the guarantee to a flow is unaffected by the behavior of other flows sharing the same server. There is no assumption that sources are flow controlled or well behaved. We first introduce and define the concept of an active flow. The delay guarantee is then formally stated as a theorem. We show how to obtain delay bounds from the delay guarantee of a single server for different specifications.

Scheduling variable-length messages in a single-hop multichannel local lightwave network

Abstract: The design of a medium access control scheme for a single-hop, wavelength-division-multiplexing-(WDM) multichannel local lightwave network poses two major difficulties: relatively large transmitter/receiver tuning overhead and large ratio of propagation delay to packet transmission time. Most schemes proposed so far have ignored the tuning overhead, and they can only schedule fixed-length packet transmissions. To overcome these two difficulties, the authors propose several scheduling algorithms which can reduce the negative impact of tuning overhead and schedule variable-length messages. A separate channel (control channel) is employed for transmission of control packets, and a distributed scheduling algorithm is invoked at each node every time it receives a control packet. By allowing the length of messages to be variable, a long message can be scheduled with a single control packet transmission, instead of fragmenting it into many fixed-length packets, thereby significantly reducing the overhead of control packet transmissions and improving the overall system performance. Three novel scheduling algorithms are proposed, varying in the amount of global information and processing time they need. Two approximate analytical models are formulated to study the effect of tuning time and the effect of having a limited number of data channels. Extensive simulations are conducted. Average message delays are compared for all of the algorithms. >

Computing time delay and its effects on real-time control systems

Abstract: The reliability of a real-time digital control computer depends not only on the reliability of the hardware and software used, but also on the time delay in computing the control output, because of the negative effects of computing time delay on control system performance. For a given fixed sampling interval, the effects of computing time delay are classified into the delay and loss problems. The delay problem occurs when the computing time delay is nonzero but smaller than the sampling interval, while the loss problem occurs when the computing time delay is greater than, or equal to, the sampling interval, i.e., loss of the control output. These two problems are analyzed as a means of evaluating real-time control systems. First, a generic analysis of the effects of computing time delay is presented along with necessary conditions for system stability. Then, we present both qualitative and quantitative analyses of the computing time delay effects on a robot control system, deriving upper bounds of the computing time delay with respect to system stability and system performance. >

Wide area mobile communication networks with multiple routing mode options

Abstract: A packet switched multi-mode mobile communication network and fixed and mobile devices for use therewith are disclosed. Each of the mobile vehicle equipment and a base station packet switch are coupled to respective data terminal equipment which generates packet data messages. The packet data messages have message characteristics associated therewith. Each of the mobile vehicle equipment and the base station packet switch have an intelligent switching node incorporated therein for selecting which of a plurality of radio frequency transmission paths to use in transmitting the packet data message to the other of the mobile vehicle equipment and the base station packet switch. Each of the plurality of radio frequency transmission paths has transmission path characteristics associated therewith. The intelligent switching nodes select the radio frequency transmission path as a function of the message characteristics of the packet data message and as a function of the transmission path characteristics of the plurality of radio frequency transmission paths.

Packet radio communication system protocol

Abstract: A method and apparatus for transmitting data in a packet radio communication system having data sources, destinations and intermediate repeaters. According to a packet protocol, a repeat count in the protocol is decremented each time a packet is retransmitted, until the repeat count reaches zero, at which time the packet is discarded. According to another packet protocol, a sequence index is used to prevent duplicate packets from being received by requiring that the sequence number fall within a sequence number window at each device, which is incremented each time a packet is received. The sequence number is also used to cause the retransmission of packets which are lost, at which time the sequence number windows in the devices which are affected are reset to allow transmission of the lost packet.

Trading packet headers for packet processing

Abstract: In high speed networks, packet processing is relatively expensive while bandwidth is cheap. Thus it pays to add information to packet headers to make packet processing easier. While this is an old idea, we describe several specific new mechanisms based on this principle. We describe a new technique, source hashing, which can provide O(1) lookup costs at the Data Link, Routing, and Transport layers. Source hashing is especially powerful when combined with the old idea of a flow ID; the flow identifier allows packet processing information to be cached, and source hashing allows efficient cache lookups. Unlike Virtual Circuit Identifiers (VCIs), source hashing does not require a round trip delay for set up. In an experiment with the BSD Packet Filter implementation, we found that adding a flow ID and a source hash improved packet processing costs by a factor of 7. We also found a 45% improvement when we conducted a similar experiment with IP packet forwarding. We also describe two other new techniques: threaded indices, which allows fast VCI-like lookups for datagram protocols like IP; and a Data Manipulation Layer, which compiles out all the information needed for Integrated Layer Processing into an easily accessible portion of each packet.

A method for transmitting packet data with hybrid FEC/ARQ type II

Abstract: 1. The invention relates to a method for transmitting packet data in an air interface of digital cellular radio telephone system based on a hybrid forward error correction/automatic repeat request, i.e. FEC/ARQ type II. comprising the following steps for each packet: a) encoding the bits of the packet of user data, header and frame check sequence, using error correcting codes and storing the resulting bits for transmission at the sender side; b) transmitting of a predefined part of the encoded bits that is bigger than the original packet; c) collecting and storing all received data at the receiver side with a storage capacity big enough to store at least the encoded bits of one packet; d) deciding on transmission success at the receiver side, based on the stored data and sending a positive acknowledge (ACK) if the transmission has been decided to be successful and a negative acknowledgement (NAK) if not; e) checking at the sender side of the positive or negative acknowledgement (ACK or NAK) and going to step b) in case of negative acknowledge (NAK) otherwise in case of positive acknowledgement (ACK) the transmitting of the packet is being completed and the receiver storage is cleared. With this method optimum throughput in response to a fluctuation in the error rate on a packet radio channel is procided.

Optical packet switching system

Abstract: An optical signal switch identifies the routing of each optical packet, directs each packet arriving at any input to the output corresponding to the routing, and includes delay lines through which the optical signals pass selectively. The switch has three stages: a first stage directing each optical packet received at an input to a chosen delay line, a second stage coupled to the output of the delay lines to direct the optical packets selectively to the output corresponding to the determined routing, and an output third stage.

Leave-in-Time: a new service discipline for real-time communications in a packet-switching network

Abstract: Leave-in-Time is a new rate-based service discipline for packet-switching nodes in a connection-oriented data network. Leave-in-Time provides sessions with upper bounds on end-to-end delay, delay jitter, buffer space requirements, and an upper bound on the probability distribution of end-to-end delays. A Leave-in-Time session's guarantees are completely determined by the dynamic traffic behavior of that session, without influence from other sessions. This results in the desirable property that these guarantees are expressed as functions derivable simply from a single fixed-rate server (with rate equal to the session's reserved rate) serving only that session. Leave-in-Time has a non-work-conserving mode of operation for sessions desiring low end-to-end delay jitter. Finally, Leave-in-Time supports the notion of delay shifting, whereby the delay bounds of some sessions may be decreased at the expense of increasing those of other sessions. We present a set of admission control algorithms which support the ability to do delay shifting in a systematic way.

Delay control for frame-based transmission of data

Abstract: A network for transferring packet data in a frame structure, preferably mixed with isochronous data. The frame structure is a continuously repeating structure, with each frame having a number of time slots. Certain ones of the time slots are available for transmitting packet data. The packet data is re-timed, e.g., by using a FIFO to output the data nibble-wise as required by the frame structure. Information about variability in delays at the transmitting end is sent to the receiving end. The receiving end uses the information to eliminate the variability, such as by a variable delay FIFO, thus restoring/recreating the original packet and IFG timing. Preferably, the frame structure is translated to and from a packet structure to permit the present invention to be used with previously available packet circuitry such as a media access controller and a hub repeater circuit.

A method and apparatus for packet alignment in a communication system

Abstract: A method and apparatus for packet alignment in a simulcast system includes a communications controller (310) having a vocoder and an input/output (I/O) processor (313) controlling the vocoder (311), the I/O processor having a timing alignment control (315) and a packet counter (314) for inserting timing alignment tags in voice/data packets and adjusting packet numbering and timing based on alignment request from a master radio communication unit. The communications controller is coupled to radio communications units (330, 340, 350), each having a packet buffer (331) and an alignment processor (332) which includes a packet number detector (333) for detecting the transmission timing tag and sending a request for a packet numbering adjustment to the communications controller and a timing alignment detector (334) for comparing a receive time of the first information packet and a predetermined preferential receive time and for sending a timing alignment request to the communications controller.

Decentralized Channel Management in Scalable Multihop Spread-Spectrum Packet Radio Networks

Abstract: This thesis addresses the problems of managing the transmissions of stations in a spread-spectrum packet ratio network so that the system can remain effective when scaled to millions of nodes concentrated in a metropolitan area. The principal difficulty in scaling a system of packet radio stations is interference from other stations in the system. Interference comes both from nearby stations and from distant stations. Each nearby interfering station is a particular problem, because a signal received from it may be as strong as or stronger than the desired signal from some other station. Far-off interfering stations are not individually a problem, since each of their signals will be weaker, but the combined effect may be the dominant source of interference. The thesis begins with an analysis of propagation and interference models. The overall noise level in the system (mainly caused by the many distant stations) is then analyzed,and found to remain manageable even as the system scales to billions of nodes. A scheme for designing a scalable packet packet radio network is then presented. Included is a method of scheduling packet transmissions to avoid collisions (caused by interference from nearby stations) without the need for global coordination or synchronization. Simulations of a system of one thousand stations are used to verify and illustrate the methods used. A method of choosing routes (minimum-energy routes) is demonstrated in simulation to produce a fully connected and functional network for one hundred and one thousand randomly placed stations. Unfortunately, congestion as the system scales is unavoidable if the traffic is not limited to some degree of locality. If traffic is limited to a few hops, then for a large system the techniques presented in this thesis are superior to ideal time division multiplexing of a clear channel.

Method of dynamically compensating for variable transmission delays in packet networks

Abstract: A method is described for playing out packets, such as voice or video packets, received through a packet network subject to variable transmission delays. The incoming packets are received in a delay buffer and a predetermined delay applied to the first packet of a sequence of packets. A variable delay is applied to subsequent packets to produce an appropriate constant play-out rate to reproduce the desired output. The fill level of the delay buffer is monitored and the predetermined delay applied to the first packet of a following sequence of packets adjusted to maintain the fill level within desired limits to minimize the risk of said buffer underflowing or overflowing.

Packet transmission system

Abstract: In a packet transmission system for transmitting a bit stream of a prescribed rate while also packeting the bit stream, a transmission side is provided with a unit for periodically providing the packeted bit stream with the position information of the head bits thereof and a circuit for adding each packet with the time of the supplied head bit position, and a reception side is provided with a temporary storage memory for temporarily storing the received packet, a data extractor for extracting the time of the head bit position which is added to the received packet, and a read-out controller for controlling the rate of a read-out operation of the bit stream from the temporary storage memory using the extracted time.

Performance of modulation-level-controlled adaptive-modulation under limited transmission delay time for land mobile communications

Abstract: The performance of an adaptive-modulation/time division duplex (TDD) scheme to support constant-bit-rate services under transmission delay time constraint conditions is evaluated in this paper. According to the received signal conditions, the modulation levels or transmission modes are selected from transmission-off, QPSK, 16QAM, or 64QAM by using the reciprocity of the propagation path characteristics in the TDD systems. Moreover, the transmission delay time is limited to keep data transmission rate constant by the memory buffer. The received signal condition is estimated by using extrapolation of pilot symbol or estimated received signal distortion at the decision stage. The performance is evaluated by computer simulation in case of 32 ksymbol/s transmission and is compared with the performance of conventional trellis-coded (TCM) 32QAM. The results show that the proposed method improves the BER performance and delay spread immunity of a system transmitting data at a constant rate under multipath fading conditions and the performance is better than that of TCM-32QAM in Rayleigh fading conditions with small Doppler frequency.

Circuits and methods for intelligent acknowledgement based flow control in a processing system network

Abstract: The present invention is directed, in general, to network connectivity, and more specifically to circuits and methods for intelligent acknowledgment-based flow control in a processing system network. The present invention concerns governing transmission of data packets and reception indicia by a transmission circuit over a network. Detector circuitry is included and is operative to (1) monitor a first latency characteristic of the network that is indicative, at least in part, of a utilization level of the network; and (2) monitor a second latency characteristic indicative, at least in part, of an efficiency level associated with transmission of the reception indicia by the transmission circuit. Control circuit is further included and is associated with the detector circuitry and the transmission circuit. The control circuit is operative to (1) adjust a retransmission delay of the transmission circuit of the data packets over the network as a function of the first latency characteristic, thereby allowing the management circuit to manage the retransmission delay as a function of the utilization level of said network; and (2) adjust a transmission delay of the transmission circuit of the reception indica over the network as a function of the second latency characteristic to thereby allow the management circuit to manage the transmission delay as a function of the efficiency level associated with the transmission circuit.

Mobile communication system with reliable handover scheme

Abstract: A mobile communication scheme realizing highly reliable handover and improved service quality. The communication signals containing identical communication data are transmitted simultaneously from the mobile station to more than one base stations at a time of the handover, such that reception signals are composed from the communication signals received by more than one base stations by using the reliability information for the received communication signals. For the packet communication, the identification information for identifying each packet is attached to each packet, and the reliability information for each packet is measured, such that the reception packets can be obtained according to the identification information attached to each packet and the reliability information measured for each packet.

Time-division multiple-access method for packet transmission on shared synchronous serial buses

Abstract: A plurality of packet data sources, eg, packet application modules, and synchronous data sources, eg, synchronous application modules, are coupled to the same TDM bus for communicating data to a network access module A portion of the bandwidth, or time-slots, of the TDM bus is allocated as a multiple-access packet channel that is shared among the packet application modules As a result, the network access module receives a single, continuously multiplexed, packet stream for transmission to an opposite endpoint Packet application modules on the TDM bus contend for this multiple access packet channel when transmitting to the opposite endpoint In the receiving direction, each packet application module accepts the entire received packet stream from the network access module and may either filter the packets using their address field or may transparently forward the data to a packet service

Multi-port network adapter

Abstract: The port in a packet network switching system that a packet should be associated with is determined by retrieving packet address information for a packet that is to be transmitted. A predetermined number of bits from the packet address information is selected to use a hash key, which is used to compute a table address. The contents of the table at that address are compared with the packet address information. If it matches, the packet is transmitted over the port associated with that particular destination address. If it does not match, the table address is incremented by one, and the contents of the new table location identified by the incremented address are compared with the packet address information. A high speed digital video network apparatus which utilizes the hashing function is implemented on a single integrated circuit chip, and includes a network protocol processing system interconnection, compression/decompression circuits, and encoder/decoder circuits.

Method for setting up virtual connections in packet switching networks

Abstract: A method for setting up virtual connections in packet switching networks is disclosed wherein a virtual connection within a second packet switching network is used to maintain or create a virtual connection between subscriber equipment in a first packet switching network. The first packet switching network employs the signaling protocol of the second packet switching network when initiating the second virtual connection so that it is not necessary for the second packet switching network to recognize a specific signaling protocol.

Bounding delays in packet-routing networks

Abstract: We consider the problem of computing the average packet delay in a general dynamic packet-routing network with Poisson input stream, during steady-state. Any packet-routing network can be formulated as a queueing network, where each server has a constant service time and the packets are served in a rst-comerst served (FCFS) order. If each server had exponentiallydistributed service time, queueing theory techniques could be used to determine the expected packet delay. However, it is not known how to compute the average packet delay for all but the simplest networks with constant time servers. It has been conjectured that to get an upper bound on expected packet delay in the constant service network, one can simply replace each constant time server with an exponential server of equal mean service time. We prove that for a large class of networks, this conjecture is true, but that there exists a network for which it is false. This large class of networks is the Markovian queueing networks. Markovian queueing networks are important because they include many packet-routing networks where the packets are routed to random destinations.

Transparent optical node structure

Abstract: A transparent optical node (TON) structure for use in a two-connected optical packet switching architecture. The TON is used in an interconnect employing deflection routing with two wavelength signaling where the data information of the packet resides on one wavelength λ 1 and the header information of the packet resides on a different wavelength λ 2 . The TON is capable of full switch functionality: packet routing, add/drop multiplexing and packet buffering. The header portion of the packet is removed from data portion and can be rewritten before reinsertion. This invention allows for the use of a previously developed TON to be used with two-wavelength signaling.