Showing papers on "Transmission delay published in 1996"

A channel access scheme for large dense packet radio networks

Abstract: Prior work in the field of packet radio networks has often assumed a simple success-if-exclusive model of successful reception. This simple model is insufficient to model interference in large dense packet radio networks accurately. In this paper we present a model that more closely approximates communication theory and the underlying physics of radio communication. Using this model we present a decentralized channel access scheme for scalable packet radio networks that is free of packet loss due to collisions and that at each hop requires no per-packet transmissions other than the single transmission used to convey the packet to the next-hop station. We also show that with a modest fraction of the radio spectrum, pessimistic assumptions about propagation resulting in maximum-possible self-interference, and an optimistic view of future signal processing capabilities that a self-organizing packet radio network may scale to millions of stations within a metro area with raw per-station rates in the hundreds of megabits per second.

Method and apparatus for switching data packets in a data network

Abstract: A local area network switch which implements packet segmentation and reassembly for cell-based switching on a backplane cell bus. A plurality of packet processing units are each coupled to a backplane cell bus wherein each packet processing unit hosts a plurality of local area network ports. Each packet processing unit is associated with a single packet buffer memory that is shared by the ports associated with the packet processing unit. The segmentation of local area network packets into fixed-size cells facilitates an efficient local area network switch which provides dedicated bandwidth for each of the ports associated with the switch. There is also provision for coupling the local area network ports to communicate with a high-speed network interface.

Method and apparatus for performance monitoring in electronic communications networks

Abstract: Management packets are defined that are modified in the payload by each node along a virtual connection and are used to measure both end-to-end QoS and specific individual intermediate node performance parameters. Management packets are implemented by defining entirely new packets or by modifying ATM OAM cells. Switches or routers for use as intermediate nodes are defined that modify the payload of the management packet and locally measure packet delay and packet loss. An intermediate node measures and records the difference between the arrival and departure times of management packets at that switch utilizing delay-stamp fields within the management packets and either the switch internal routing header or timestamp fields within the packet. At the endpoint of the virtual connection, delay-stamp fields in the management packet indicate individual node delays and the cumulative delay. An intermediate node counts the number of packets it discards and records these values in the payload of the management packet individually and cumulatively.

Packet radio system and methods for a protocol-independent routing of a data packet in packet radio networks

Abstract: The present invention relates to a protocol-independent routing of data packets between a mobile station of a packet radio network and a party (Host) connected to an external network. In the invention, a data packet of an extraneous protocol (IPX) is transferred through a packet radio network using a second protocol (X.25) as encapsulated in a data packet according to the second protocol. The transferring packet radio network does not thus need to understand the protocol of the transferred extraneous data packet or to be able to interpret the contents of the data packet. A data packet network is connected to other packet radio networks, data networks or the backbone network between packet data networks via a gateway node (GPRS GSN), which uses the network-internal protocol (X.25) towards the dedicated packet network and the protocol of each network towards other networks. When a data packet is transferred via a gateway node from a network into another network, the data packet is encapsulated in a packet according to the protocol of the new network. When the encapsulated data packet arrives in a node which supports the protocol of the encapsulated data packet, the encapsulation is stripped away and the data packet is routed forward according to the protocol of the data 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 identifier (flow ID); the flow ID 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 seven. 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 (DML), which compiles out all the information needed for integrated layer processing (ILP) and scheduling into an easily accessible portion of each packet.

Adaptive error control for packet video in the Internet

Abstract: Anecdotal evidence suggests that the quality of many videoconferences in the Internet is mediocre because of high packet loss rates. This makes it important to design and implement mechanisms that minimize packet loss and its impact in video (and audio) applications. There are two such types of mechanisms. Rate control mechanisms attempt to minimize the amount of packet loss by matching the bandwidth requirements of a video flow to the capacity available in the network. However, they do not prevent packet loss altogether. Error control mechanisms attempt to minimize the visual impact of lost packets at the destinations. We provide motivation for using error control mechanisms based on forward error correction (FEC) and packet reconstruction. We examine a specific mechanism, and evaluate its cost as well as the benefit expected from using it. This mechanism can be augmented to obtain a joint source/channel coding scheme suitable for both the current and the future integrated services Internet.

Method and apparatus for regulating data flow in networks

Abstract: Interpacket delay times are modified in full-duplex Ethernet network devices by calculating for each network station a delay interval based on a time to transmit a data packet at the network rate and a calculated time to transmit the data packet at a desired transmission rate. The network station waits the calculated delay time following a packet transmission before transmitting the next data packet, ensuring that the overall output transmission rate of the network station corresponds to the assigned desired transmission rate. The desired transmission rate is received as a media access control (MAC) control frame from a network management entity, such as a switched hub. Hence, each network station operates at the desired transmission rate, minimizing the occurrence of congestion and eliminating the necessity of PAUSE frames.

Packet transmission node device realizing packet transfer scheme and control information transfer scheme using multiple virtual connections

Abstract: A packet transmission node which realizes a packet transfer scheme in which a plurality of virtual connections for different qualities of service are set up in correspondence to a multicast destination address, and output virtual connection identifiers are stored in correspondence to destination addresses in a routing table, so that a packet is transferred to output virtual connections determined by referring to the routing table according to a destination address of a packet. A packet transmission node also realizes a control information transfer scheme in which different output virtual connections are set up for a user data packet and a control packet having an identical destination address, and output virtual connection identifiers are stored in correspondence to destination addresses and upper layer protocol identifiers in a routing table, so that a packet is transferred to an output virtual connection determined by referring to the routing table according to a destination address and an upper layer protocol identifier of a packet.

Throughput analysis of DS/SSMA unslotted ALOHA system with fixed packet length

Abstract: Throughput analysis of direct-sequence spread spectrum multiple access (DS/SSMA) unslotted ALOHA with fixed packet length is presented. As the levels of multi-user interference fluctuate during the packet transmission, we calculate the packet error probability and the throughput by considering not only the number of overlapped packets but also the amount of time overlap. On the assumption that packet generation is Poisson, the system can be thought as the queueing system M/D//spl infin/. With Gaussian approximation of multi-user interference, we obtain the throughput as the function of the number of chips in a bit, the packet length, and the offered load of the system. We also analyze the channel load sensing protocol (CLSP), and obtain the optimum threshold of CLSP.

PRMA/DA: a new media access control protocol for wireless ATM

Abstract: We propose a media access control (MAC) protocol for wireless local area networks (LANs) that is capable of supporting various types of traffic demands, such as constant bit rate (CBR) voice, variable bit-rate (VBR) video, and packet data. In addition, the proposed protocol provides a seamless connectivity to a broadband ATM backbone network. Our protocol, having an air interface comparable to ATM, adopts a dynamic channel allocation scheme which enables expeditious network-access and utilizes bandwidth resource efficiently. The simulation results presented show the improvements of dynamic channel allocation over the static channel allocation scheme in terms of key performance metrics such as: throughput, call blocking probability, network access delay and cell transmission delay.

Distributed packet switching in arbitrary networks

Abstract: In a seminal paper Leighton, Maggs, and Rao consider the packet scheduling problem when a single packet has to traverse each path. They show that there exists a schedule where each packetreaches its destination in O(C + D) steps, where C’ is the congestion and D is the dilation. The proof relies on the Lovbz Local Lemma, and hence is not algorithmic. In a followup paper Leighton and Maggs use an algorithmic version of the Local Lemma due to Beck to give centralized algorithms for the problem. Leighton, Maggs, and Rao also give a distributed randomized algorithm where all packets reach their destinations with high probability in O(C’ + D log n) steps. In this paper we develop techniques to guarantee the high probability of dehvering packets without resorting to the Lovi$sz Local Lemma. We improve the distributed algorithm for problems with relatively high dilation to O(C) + (log* n)”(iog” ‘JD -t poly(log n). We extend the techniques to handle the case of infinite streams of regularly scheduled packets along every path. Here we measure the congestion on an edge e by the sum of the rates of the packet streams that use the edge, denoted by A(e). We require that for some small constant e > 0, for every edge e, J(e) S 1 – c. In this case we use the parameter R = mw ri, the maximum distance between packets of the same stream, instead of the congestion C above. We notice that max{R, D} is a worst case lower bound on the maximum delay of a packet. We also extend the results to a model of packet traf*Supported in part by the NSF PYI award of $va Tardos. Part of this work was performed while visiting the School of OR&IE at Cornell, and while a postdoctoral fellow at the University of Toronto Computer Science Department. Work at the Technion supported in part by the Ruth and David Moskowitz Academic Lectureship award. t Research supported in part by a Packard Fellowship and ZUI NSF PYI award, by NSF through grant DMS 9505155, and ONR through grant NOO014-961-0050. Permission to make digitel/bard copies of all or pad of this material for personal or classroom use is granted without fee provided that the copies are not made or distributed for profit or conunereial advantage, the copyright notice, the title of the publication and its date appear, and notice is given that copyright is by permission of tie ACM, Inc. To c~y otherwise, to republish, to peat on servers or to redlatribute to lists, requmes specific permission and/or fee. STOC’96, Philadelphia PA, USA @ 1996 ACM 0-89791-785.5/96/05. .$3.50 fic for handling bursty communication. The model is motivated by the new adversarial model suggested by Borodin et al.

Discovery of acceptable packet size using ICMP echo

Abstract: A network messaging protocol is used to send an arbitrary size packet over the network from a source machine to a destination machine. If the packet is accepted by the routers along the path of the potential network connection, then the destination machine will be able to echo the test message packet back to the source machine and an acceptable packet size is set equal to the test message packet size. If not, after a suitable time out, the source machine will send another test message with a different packet size than the initial arbitrary packet size and wait for echo and repeat the process until an acceptable packet size is discovered.

Adaptive methods for packet transmission over wireless networks

Abstract: A method for transmission of digital information packets over a transmission channel. A transmitter computes a current value of a carrier-sense defer threshold and a transmit power level, based an observed ratios between recent packet transmission successes, recent packet collisions, recent packet deferrals, and an estimate of a path-loss characteristic of the transmission channel. The computed power level and carrier-sense defer threshold are selected to obtain a desired ratio of at least two of future packet success, future packet collisions, or future packet deferrals. The current carrier-sense defer threshold value is used to determine when a carrier signal for another transmitter's digital information packet is present on the transmission channel; transmission is deferred until the channel is determined to be clear of other packets. The packet is then transmitted at the computed power level.

Impact of tuning delay on the performance of bandwidth-limited optical broadcast networks with uniform traffic

Abstract: This paper studies the effects of tuning delay of transmitters in packet-based optical broadcast networks. We consider scheduling of random traffic with tunable transmitters and fixed-tuned receivers and obtain the degradation imposed by tuning delay using several performance criteria, such as schedule completion time, average packet delay, and session blocking rates. We show that for off-line scheduling the effects of tuning delay are small even if the tuning time is as large as the packet duration. We provide a lower bound to the expected completion time of any off-line schedule with an arbitrary number of wavelengths. We then describe a near-optimal schedule which is based on the principle of having idle transmitters tune to wavelengths just-in-time to start their transmissions. Stability and capacity issues in the transmission of real-time traffic are considered and a queueing-theoretic analysis of average packet delay is given. The packet delay is found to be insensitive to tuning delay under near-optimal transmission scheduling. Finally we extend the model to connection-oriented networks and evaluate the session blocking performance for scheduled circuit connections.

Maintaining high throughput during overload in ATM switches

Abstract: This paper analyzes two popular heuristics for ensuring packet integrity in ATM switching systems. In particular, we analyze the behavior of packet tail discarding, in order to understand how the packet level link efficiency is dependent on the rates of individual virtual circuits and the degree of the imposed overload. In addition, we study early packet discard and show that the queue capacity needed to achieve high efficiency under worst-case conditions grows with the number of virtual circuits and we determine the efficiency obtainable with more limited queue capacities. Using the insights from these analyses, extensions to early packet discard are proposed which achieve high efficiency with dramatically smaller queue capacities (independent of the number of virtual circuits).

Method and system for increasing network information carried in a data packet via packet tagging

Abstract: A method and system for providing statistical network information carried in a data packet (8) being transmitted on a network. The method includes the steps of receiving a data packet having a data portion on a repeater (12) and transferring the data portion to a management unit (10). The method further includes the step of appending statistical information to the data portion during an inter-packet gap period. The apparatus for increasing information in a data packet on a network includes a repeater mechanism, a management unit mechanism, and a packet tagging circuit. The repeater mechanism receives a data packet having a data portion, the management unit mechanism determines statistical information based on the data packet, and the packet tagging circuit appends information to the data portion of the data packet during an inter-packet gap period.

Flow control method and apparatus for ethernet packet switched hub

Abstract: A flow control enabled Ethernet switch that applies backpressure to input ports that attempt to transmit a data packet to a busy destination port. The backpressure is a phantom packet that activates the IEEE 802.3 collision detection system to abort transmission of the data packet. A subsequent series of phantom packets (that are invalid) provide carrier activity that inhibits retransmission of the packet. When the destination port is available, the phantom packets are stopped, permitting standard retransmission of the data packet. The switch includes prioritization mechanisms (e.g., a throttle count) to use when awarding priority to a port having backpressure applied.

Arrangement for regulating packet flow rate in shared-medium, point-to-point, and switched networks

Abstract: Delay times are modified in full-duplex Ethernet network devices by calculating in each network station a delay interval based on a size of a transmitted data packet and a desired transmission rate. The network station waits the calculated delay time before transmitting another data packet, ensuring that the overall output transmission rate of the network station corresponds to the assigned desired transmission rate. The desired transmission rate is received as a media access control (MAC) control frame from a network management entity, such as a switched hub. Hence, each station operates at the desired transmission rate, minimizing the occurrence of congestion and eliminating the necessity of PAUSE frames.

Throughput performance of SAW-ARQ protocol with adaptive packet length in mobile packet data transmission

Abstract: In a noncellular or large cell-size mobile radio communication system, log-normal shadowing as well as Rayleigh fading becomes the predominant source of system degradation. This paper proposes an efficient stop and wait automatic repeat request (SAW-ARQ) protocol with adaptive packet length to provide reliable mobile data packet transmission. The adaptive SAW-ARQ protocol controls the transmitting packet length according to the time-varying channel condition estimated with the number of ACK (acknowledgment packet) and NACK (negative-acknowledgment packet). Computer simulation results show that the proposed protocol can achieve high throughput and reduce the number of retransmission effectively for slow and fast Rayleigh fading/log-normal shadowing conditions.

Traffic flow regulation to guarantee end-to-end delay in packet switched networks

Abstract: The present invention relates to the issue of providing end-to-end delay guarantees in a multi-node communication system. More specifically, the present invention addresses the problem of specifying operational parameters of rate-controlled service disciplines in a communication network in order to efficiently provide end-to-end delay guarantees. The key contribution is a method for specifying leaky bucket parameters as well as scheduling delays at each node, which are used as inputs to the rate-controlled service discipline.

Method and apparatus for packet scheduling using queue length and connection weight

Abstract: A packet scheduling scheme capable of realizing a fair scheduling regardless of weights of connections. A packet scheduler has a plurality of packet queues for temporarily storing entered packets, to each of which a weight is set up; a packet input unit for entering packets into the packet queues; a scheduling information management unit for managing scheduling information for specifying an order to read out packets stored in the packet queues, according to a queue length of each packet queue and the weight set up for each packet queue; and a packet output unit for reading out and outputting desired packets from the packet queues according to the scheduling information. In a case of fixed length packets, the scheduling information management unit always holds as many scheduling information corresponding to one packet queue as a smaller one the queue length of that one packet queue and the weight set for that one packet queue, so as to make a processing time required for scheduling constant regardless of a number of connections.

Optimizing packet size to eliminate effects of reception nulls

Abstract: Packets are transmitted in different block sizes (21, 26) according to the speed of motion of the receiver (24, 29). The packet block size (21, 26) is selected to minimize the effects of burst errors that occur at the receiver. The burst errors are disbursed between all packets in the packet block by interleaving the packets together prior to transmission. The receiver then deinterleaves the packets into their original format disbursing burst errors between all packets in the packet block. Since each packet will only contain a small proportion of the burst error, standard ECC schemes can be used to correct for bit errors in each packet increasing the probability that all packets will be transmitted successfully.

Data driven information processor

Abstract: A data driven processor includes an output processing unit outputting a data packet outside the processor while referencing a branch control parameter register group. In the register group, a processor number/generation number specifying parameter P/G, a branch comparison parameter RM, and a branch comparison data parameter RD are stored. At the time of output of the data packet, the output processing unit reads out any one of a processor number and a generation number in the data packet according to parameter P/G, and sends out the data packet to any one of output ports OA and OB according to the result of predetermined operation processing using the read out number and parameters RM and RD. When a plurality of processors which operate as described above are connected to each other to carry out processing simultaneously while inputting/outputting a data packet, a data path among the processors can be set and changed easily according to any of the processor number and the generation number in the packet.

Cable modem interface unit for capturing and processing incoming packets from a cable modem

Abstract: A cable modem interface unit is positioned between a cable modem and a network driver interface layer. The cable modem receives packets from a packet source. The interface unit includes a control packet filter coupled to the modem. The control packet filter receives a packet from the cable modem and determines whether the packet is a control packet or a data packet. The interface unit further includes a receive unit coupled to the control packet and the network driver interface layer. If the control packet filter determines that the packet is a data packet, the receive unit receives the packet from the control packet filter and sends the packet to the network driver interface layer. The interface unit further includes a protocol handler coupled to the receive unit. If the control packet filter determines that the packet is a control packet, the protocol handler receives the packet from the control packet filter.

Scalable parallel packet router

Abstract: A scalable parallel packet router comprises a massively parallel computer (10) and a plurality of multiplexers (15), and is controlled by a disclosed packet routing algorithm, and a method of very high-speed packet routing. The method includes selection of a destination node by computing a hash function from the source and destination addresses of a packet, so the ordering of packets can be preserved, while a statistically uniform load distribution among components within the packet router can be achieved.

A new error concealment technique for audio transmission with packet loss

Abstract: We present a new error concealment technique for audio transmission over packet networks with high packet loss rate. Unlike other techniques it modifies the time-scale of correctly received packets instead of repeating them. This is done by a time-domain algorithm, WSOLA, whose parameters are redefined so that short audio segments like lost packets can be extended. Particular attention is paid to the additional delay introduced by the new technique. For subjective hearing tests, single and double packet loss is simulated at high packet loss rates, and the new technique is compared to previous proposals by category judgment and component judgment of sound quality. Mean Opinion Score (MOS) curves show that sound distortions due to packet repetition can be reduced.

Minimum-latency data mover with auto-segmentation and reassembly

Abstract: A data mover which provides guaranteed transfer of data between two locations. The data mover includes a pair of data packet memories for input, a pair of data packet memories for output, and a controller which alternately switches each of the paired data packet memories between a data loading mode and a data unloading mode. The controller enables one of the paired data packet memories in the data loading mode and enables the other one of the paired data packet memories in the data unloading mode. The controller switches the modes of the paired data packet memories upon receiving an acknowledgement of moved data. By (a) switching the paired data packet memories upon receiving the acknowledgement, (b) segmenting the data packets (enabling data transfer initiation before all source data becomes available) and/or (c) embedding a response (enabling transmission of the next packet without first waiting for complete transmission of an incoming packet), complete transmission of the bi-directional data is automatically segmented and reassembled to minimize latency.

Packet multiplexing transmission apparatus

Abstract: A sender apparatus in a packet multiplexing transmission apparatus for packetizing lengthy information supplied from a plurality of information channels for multiplex transmission. Each packet is affixed with a time stamp for establishing synchronization in operation between the sender and a receiver. After multiplexing the data for transmission, packets are generated and then affixed with the respective time stamps in order to make constant a delay time which intervenes between the affixation of the time stamp to the packet and the send-out thereof onto a network.

A terabit per second packet switch having assignable multiple packet loss probabilities

Abstract: A physically realizable one terabit or more ATM packet switch that has a large number of input interfaces connected to a single stage switching fabric which is in turn connected to a number of output modules, generally according to the growable packet switch architecture. This ATM packet switch is different from other growable packet switches in that it has a single stage switch fabric controlled by an out-of-band controller, yet it has significantly reduced complexity with respect to comparably sized electronic crossbar switches or their isomorphs. The out-of-band controller has multiple priority levels in order to provide high priority users with a near certainty that their packets will be successfully routed, while delivering an acceptably low packet or cell loss probability to users at the lowest priority level.