Showing papers on "Transmission delay published in 1993"

End-to-end packet delay and loss behavior in the internet

Abstract: We use the measured round trip delays of small UDP probe packets sent at regular time intervals to analyze the end-to-end packet delay and loss behavior in the Internet. By varying the interval between probe packets, it is possible to study the structure of the Internet load over different time scales. In this paper, the time scales of interest range from a few milliseconds to a few minutes. Our observations agree with results obtained by others using simulation and experimental approaches. For example, our estimates of Internet workload are consistent with the hypothesis of a mix of bulk traffic with larger packet size, and interactive traffic with smaller packet size. We observe compression (or clustering) of the probe packets, rapid fluctuations of queueing delays over small intervals, etc. Our results also show interesting and less expected behavior. For example, we find that the losses of probe packets are essentially random unless the probe traffic uses a large fraction of the available bandwidth. We discuss the implications of these results on the design of control mechanisms for the Internet.

System for securing inbound and outbound data packet flow in a computer network

Abstract: A filter module allows controlling network security by specifying security rules for traffic in the network and accepting or dropping communication packets according to these security rules. A set of security rules are defined in a high level form and are translated into a packet filter code. The packet filter code is loaded into packet filter modules located in strategic points in the network. Each packet transmitted or received at these locations is inspected by performing the instructions in the packet filter code. The result of the packet filter code operation decides whether to accept (pass) or reject (drop) the packet, disallowing the communication attempt.

Characterizing End-to-End Packet Delay and Loss in the Internet

Abstract: We use the measured round trip delays of small UDP probe packets sent at regular time intervals to characterize the end-to-end packet delay and loss behavior in the Internet. By varying the interval between probe packets, it is possible to study the structure of the Internet load over different time scales. In this paper, the time scales of interest range from a few milliseconds to a few minutes. Our observations agree with results obtained by others using simulation and experimental approaches. For example, our estimates of Internet workload are consistent with the hypothesis of a mix of bulk traffic with larger packet size, and interactive traffic with smaller packet size. The interarrival time distribution for Internet packets is consistent with an exponential distribution. We also observe a phenomenon of compression (or clustering) of the probe packets similar to the acknowledgement compression phenomenon recently observed in TCP. Our results also show interesting and less expected behavior. For example, we find that the losses of probe packets are essentially random when the probe traffic uses a small fraction of the available bandwidth.

Rate-controlled static-priority queueing

Abstract: The authors propose a service discipline, called the rate-controlled static-priority (RCSP) queuing discipline, that provides throughput, delay, delay jitter, and loss free guarantees in a connection-oriented packet-switching network. The RCSP queuing discipline avoids both time-framing and sorted priority queues; it achieves flexibility in the allocation of delay and bandwidth, as well as simplicity of implementation. The key idea is to separate rate-control and delay-control functions in the design of the server. Applying this separation of functions results in a class of service disciplines of which RCSP is an instance. >

Feedback control of congestion in packet switching networks: the case of a single congested node

Abstract: Addresses a rate-based feedback approach to congestion control in packet switching networks where sources adjust their transmission rate in response to feedback information from the network nodes. Specifically, a controller structure and system architecture are introduced and the analysis of the resulting closed loop system is presented. Conditions for asymptotic stability are derived. A design technique for the controller gains is developed and an illustrative example is considered. The results show that, under appropriately selected control gains, a stable (nonoscillatory) operation of store-and-forward packet switching networks with feedback congestion control is possible. >

Traffic management in packet communications networks

Abstract: In a packet communications network, the addition or deletion of a connection to the network by a user is governed by a link traffic metric which represents the effective capacity of each link in the network which participates in the packet connection route. The link metric is calculated in real-time and updated by simple vector addition or subtraction. Moreover, this link metric is also used to calculate leaky bucket parameters which govern the access of packets to the network once the connection is set up. A packet network using these link metrics and metric generation techniques provides maximum packet throughput while, at the same time, preserving grade of service guarantees.

Method and apparatus for optimum path selection in packet transmission networks

Abstract: A packet communications system utilizes a route determining mechanism by identifying principal paths between the source and the destination in the system Principal paths are minimum hop count paths with a transmission delay less than a specified threshold Principal path links are accepted as legs of the optimum path, if feasible, ie, if the resulting load on the link is less than a specified principal threshold Secondary links are accepted only if the resulting load on the link is less than a specified secondary threshold, where the secondary threshold is less than the principal threshold All paths must also have a transmission delay less than a specified threshold Each request for a route includes the source node, the destination node, the load required, the maximum transmission delay and, if desired, the quality of service parameters which all of the legs of the route must satisfy A modified Bellman-Ford breadth-first search algorithm is used to identify the principal links and, using these principal link identifications, determining the optimum path

Packet-based data compression method

Abstract: A data processing system is described that employs data packets which include at least static and dynamic fields, the static fields containing information that often remains constant during a multi-packet communication interval and the dynamic fields containing information that changes for each packet. Many packets also include a user-data fields. A compression method is described which comprises: reformatting each data packet by associating its static fields with a first packet region and its dynamic fields with a second packet region. The process then assembles a static table that includes static information from at least an initial data packet's first packet region. It then identifies static field information in a subsequent data packet's first packet region that is common to the information in the static table. Such common information is encoded so as to reduce its data length. The common static information is then replaced in the modified data packet with the encoded common static information and the modified data packet is then transmitted. A similar action occurs with respect to user-data information. A single dictionary table is created for all packet headers, while separate dictionary tables are created for each user-data portion of a packet-type experienced in the communication network thereby enabling better compression.

High bandwidth packet switch.

Abstract: A fast packet switch comprising one buffer directly connected between a plurality of input ports and a plurality of output ports to effect rapid throughput of data packets. A pointer to a location in the buffer is allocated by a buffer manager upon receipt of notification of an incoming packet at the receiving input port and the input port delivers the packet as it is received to the location designated by the pointer. After the data packet is received, the input port delivers the pointer and a destination address for the packet to a router, which selects one of the plurality of output ports based on the destination address. The router queues the pointer in a queue for the selected output port. The output port then retrieves the data packet from the buffer using the pointer to determine the location, and transmits the data packet. After the transmission is complete, the output port returns the pointer to the buffer manager. This packet switch may be pipelined to receive, route, and transmit simultaneously on adjacent data packets.

Frame-based transmission of data

Abstract: A network for transferring packet data in a frame structure, preferably mixed with isochronous data is provided. 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. Similar re-timing can be used for isochronous data so that the frame structure defines time-division multiplexing of the packet data and isochronous data. A four/five encoding scheme provides sufficient encoding efficiency that both the packet data and other data can be accommodated without degrading the data rate of the packet data. The encoding scheme provides extra symbols which can be used for transferring "no carrier" information, or "frame alignment" messages. 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. Latency of the FIFO can be reduced by pre-filling with packet preambles, and/or sub-latency propagation of preamble bytes, or providing special MACs which do not output preambles, and using the buffer circuitry to output preambles.

Low-complexity multiple access protocols for wavelength-division multiplexed photonic networks

Abstract: Media access control protocols for an optically interconnected star-coupled system with preallocated wavelength-division multiple-access channels are discussed. The photonic network is based on a passive star-coupled configuration in which high topological connectivity is achieved with low complexity and excellent fault tolerance. The channels are preallocated to the nodes with the proposed approach, and each node has a home channel it uses either for data packet transmission or data packet reception. The performance of a generalized random access protocol is compared to an approach based on interleaved time multiplexing. Semi-Markov analytic models are developed to investigate the performance of the two protocols. The analytic models are validated through extensive simulation. The performance is evaluated in terms of network throughput and packet delay with variations in the number of nodes, data channels, and packet generation rate. >

Packet network and method for congestion avoidance in packet networks

Abstract: Packet transfer is controlled by using an acceleration rate of packet transfers or by using a packet transfer rate acceleration ratio to predict that congestion will occur at a prescribed time in the future. Congestion avoidance in packet integrated networks is thereby achieved in a network having both variable rate terminal nodes and fixed rate terminal nodes. A future packet transfer rate is predicted in a congestion prediction circuit on the basis of a pre-established upper limit for the packet transfer acceleration or acceleration ratio. When it is predicted that the packet transfer rate will exceed a permissible value, a congestion prediction signal is output or a rate increase request indication is deleted. The invention prevents packets from being discarded in the packet network, allows buffer memory capacity of nodes in the network to be decreased, and avoids the generation of new packets when signal congestion is predicted.

Contributions toward real-time services on packet switched networks

Abstract: Packet switched communications services with real-time delay constraints, such as voice and video, combine the established fields of digital signal processing and data communications networking. Each field is outlined, and new open problems due to the combination are identified. A simulation study investigates layered coding using a packet voice Markov chain source model. Information is partitioned into two or more priority layers to protect important components from loss. A parameter, $\alpha$, identifies the proportion of traffic placed in each priority. Packet loss rates for each priority and $\alpha$ are used to compute the signal to noise ratio, which is a more appropriate performance measure for voice and video services than loss rates alone. Dynamic feedback control of the priority partition based on network load conditions is shown to be effective, even with substantial feedback delay. Feedback, in conjunction with priority, provides graceful service degradation with increasing load and loss rate. A queueing analysis of this system is also investigated using a two-dimensional Markov chain source model that represents both load and $\alpha$, which vary dynamically. The simulation and analytic models are compared. Two methods for reducing the numerical complexity are given. A two-hour long empirical sample of variable rate video is derived by applying a simple intraframe video compression code to an action movie. Statistical characteristics are measured, including an accurate model for the heavy-tailed marginal distribution of video frame bandwidth. A statistical property called long-range dependence is described, measured, and shown to be significant for this data. A new traffic source model results from combining the marginal distribution with long-range dependence. Extensive trace driven simulations characterize network queueing behavior and allocation of bandwidth/buffer resources. Statistical multiplexing gain of variable rate video is evaluated as well as the advantage due to multiplexing video with data services. We discuss the implications of this traffic analysis for for the design of congestion control mechanisms for integrated packet networks. We close with some comments on the ramifications of advancing electronic hardware speed and complexity for multi-media communications.

Method and system for the arrangement of vocoder data for the masking of transmission channel induced errors

Abstract: A method and system (16) by which parameter data representative of vocoded speech are organized into a data packet for transmission so as to reduce the impact of transmission channel induced errors on the data packet. A data packet is constructed with certain most perceptually significant bits of parameter data at the beginning of the data packet. Following in the data packet are lesser perceptually significant bits of the same parameter data. Other parameter data then follows in the data packet. Interleaved in the data packet following the most perceptually significant bits at the beginning of the data packet are most perceptually significant bits of other parameter data. A parity check code is computed from the most perceptually significant bits in the data packet and also interleaved in the data packet following the most perceptually significant bits at the beginning of the data packet.

Addressing technique for storing and referencing packet data

Abstract: A hierarchical addressing technique is employed in a packet communications system to enhance flexibility in storing and referencing packet information. This method permits packet message data and certain packet control data to be stored in memory locations without having to be duplicated at a different memory location prior to transmission of the packet. This method is preferably employed in a ring configuration in which a series of packets have addressing mechanisms which points sequentially to each other to form a ring of packets received or to be transmitted.

On per-session end-to-end delay distributions and the call admission problem for real-time applications with QOS requirements

Abstract: A crucial problem facing the designers and deployers of future high-speed networks is providing applications with quality of service (QOS) guarantees. For soft real-time applications, which are delay sensitive but loss tolerant, delay distribution is an important QOS measure of interest. In this paper we study (through simulation) the end-to-end delay distribution seen by individual sessions under simple first-come first-served (FCFS) multiplexing in a network model with two significant features: (1) all traffic is connection-oriented, (2) cross traffic along routes is representative of that seen by calls in a moderately sized wide area network (i.e., less than 100 switches). We compare these delay distributions with the worst case point-valued analytic delay bounds predicted by three different techniques for providing such bounds (two of which require a more sophisticated link-level scheduling policy). We also consider the per-hop delay distributions seen as a session progresses "deeper" into the network and determine the sensitivity of these delay distributions to the manner in which the interfering traffic is modeled. Finally, we use our delay distribution results to examine the tradeoff between the QOS requested by a call, the manner in which the QOS guarantee is provided, and the number of calls that are admitted at the requested QOS.

Programmable delay line programmable delay circuit and digital controlled oscillator

Abstract: A programmable delay line comprises a plurality of delay stages connected in series, each of the delay stages including: a basic path for passing an input signal; a delay path for passing the input signal with a predetermined delay time; and a selector for selecting either the basic path or the delay path to pass the input signal in accordance with digital data externally inputted, wherein differences in times for passing the input signal through the basic path and through the delay path in the plurality delay stages are UD.2n (n=0, 1, 2 . . . ), UD being unit delay time. A programmable delay apparatus comprises: an oscillator and counter, which determine a coarse delay time in accordance with the upper bit data of control data, and a programmable delay line, which determines a fine delay time according to the lower bit data of the control data after the finish of the coarse delay time to obtain a total delay time. The counter provides a wide range of available delay times. The oscillator of the programmable delay apparatus can be controlled by a control signal. Addition of a feedback circuit for supplying the delay signal from the delay line as the control signal to the oscillator of the programmable delay apparatus provides a digital controlled oscillator.

Initializing terminals in a signal distribution system

Abstract: The invention relates to signal distribution systems, such as interactive cable television systems, wherein a central server communicates with remote terminals wherein the server assigns a specific address to each newly-connected terminal for control messages and polls. In the case of an interactive cable television system, such terminal is the set-top box or converter connected between the cable system and the television receiver. Periodically, the server broadcasts a control message containing a tentative address, and polls the tentative address. Upon receiving such a message, a newly-connected converter stores the tentative address and responds to the poll with an uplink message. After receiving the uplink message, the server inserts the tentative address in its polling list in accordance with the transmission delay measured from sending the poll to receiving the uplink message and selects a new tentative address to use in subsequent broadcast control messages. Uplink messages are echoed by the server and verified by the converter. If the converter cannot verify an echoed uplink message, which can occur if more than one newly-connected converter responds to the same broadcast control message, the converter repeats its initialization procedure to acquire a new address. After the initialization is complete, system software can be downloaded from the server to the converter and the converter placed in service.

Shared-memory optical packet (ATM) switch

Abstract: Previously, we determined fundamental performance limitations associated with `all-optical' packet switches, in which the packet buffering is implemented via fiber delay lines. In this work, we propose and analyze an optical packet (ATM) switch architecture that comes close to achieving the optimal performance (i.e., best possible delay-throughput performance and minimal possible buffer requirements) of a random-access, shared-memory design. The proposed Shared-Memory Optical Packet (SMOP) Switch buffers packets in recirculation delay lines of appropriately-selected lengths, and uses a novel control algorithm that: (i) keeps packets in their proper first-in, first-out sequence, (ii) supports multiple levels of priority traffic, (iii) minimizes the needed number of recirculation loops (which reduces the size of the switch fabric), and (iv) ensures that packets pass through the recirculation delay lines only a small number of times (e.g., less than 10).© (1993) COPYRIGHT SPIE--The International Society for Optical Engineering. Downloading of the abstract is permitted for personal use only.

Packet switch to provide code division, multiple access cellular service

Abstract: A system for relaying CDMA packetized data from a cell site or sites to a destination speech processor. The CDMA packetized data, is received at the cell site and is sent in its packetized form along with a destination address to a packet switch. A packet handler at the packet switch receives the CDMA data packets and forwards them on a packet bus. If the destination speech processor assigned to that call is connected to the packet bus, it recognizes its own address and processes the data packet.

Method and system for real-time burstiness analysis of network traffic

Abstract: Method, for use with a communications network, for measuring and analyzing the burstiness of network traffic based on the ratio of packet interarrival times. The network traffic includes at least one data burst, and each data burst includes a stream of data packets for travel over the network. The method includes (a) receiving a first data packet of the at least one data burst for travel over the network, the first data packet having an associated first packet interarrival time, and (b) consecutively receiving a second data packet for travel over the network, the second data packet having an associated second packet interarrival time. The method also includes (c) determining the ratio of the second packet interarrival time to the first packet interarrival time and (d) comparing the ratio of the second packet interarrival time and first packet interarrival time to at least one predetermined constant. Lastly, the method includes (e) determining whether the second data packet belongs to the at least one data burst based on the comparison, and (f) accumulating the second data packet with a data burst based on the comparison. A system for carrying out the method is also provided.

Methods and devices for prioritizing in handling buffers in packet networks

Abstract: Methods and devices are proposed for handling a buffer (11) in packet networks are proposed, particularly in regard of loss and delay of packets. The packets are supposed to belong to predetermined or implicitly given loss priority classes and delay priority classes. When a packet arrives to the buffer (11) the class of the packet is determined, both for loss and delay. For each loss priority class there is a predetermined threshold value (T1, TBusy, TIdle, T3, TL) and the total filling level (M) of the buffer, i.e. the total number of packets stored, is compared to the threshold value of the loss priority class to which the received packet belongs. If said threshold value (T1, TBusy, TIdle, T3, TL) is larger than said filling level (M), the packet is buffered in order to be forwarded, and otherwise it is lost. In the determination if said packet is to be buffered or lost the delay priority of the packet is not taken into account. In the forwarding of packets from the buffer (11) those packets are chosen in the usual way, which belong to higher delay priority classes before packets belonging to lower delay priority classes, where this is performed in such a way that packets belonging to all delay priority classes can be warranted a specific minimum service level. It is achieved by associating each such class with a maximum time period, within which at least one packet of the delay class considered will be forwarded, if such a packet is available in the buffer (11).

Delay Minimization For Zero-skew Routing

Abstract: a1 orithm achieve 50% reductiqn of the delay time on a tenchmark data with 3000 pins.

All digital on-the-fly time delay calibrator

Abstract: An on-chip digital servo scheme for providing continuous calibration of integrated circuit on-chip time delay devices to provide real-time regulation against various parameter or environmental changes, such as processing, temperature and power supply variations. The techniques are particularly useful for semiconductor delay lines comprised of a selectable number of identical unit delay circuits having the same propagation time. This scheme constantly monitors the delay changes in the unit delay elements and calibrates the delay line by comparing the delay against a stable, crystal controlled reference clock period to determine, in each instance, how many unit delay elements in the delay line is needed to effectively delay the amount of time to equal the reference clock period. A real time digital pointer number is generated and updated while the device is in operation. This pointer can be used to inform or update other delay or time bases on the same integrated circuit substrate which are constructed from the same type of unit delay cell and which may choose to use a different number of unit delay cells. Accordingly, this scheme can then be used for various delay regulation purposes in and all-digital circuit, such as clock multiplication, pulse width regulation, and other applications where accurate, regulated, digital command controlled delay or time bases are needed.

Providing quality of service in packet switched networks

Abstract: Increases in bandwidths and processing capabilities of future packet switched networks will give rise to a dramatic increase in the types of applications using them. Many of these applications will require guaranteed quality of service (QOS) such as a bound on the maximum end-to-end packet delay and/or on the probability of packet loss. This poses exciting challenges to network designers. In this paper we discuss the QOS requirements of different applications and survey recent developments in the areas of call admission, link scheduling, and the interaction between the provision of QOS and call routing and traffic monitoring and policing. We identify what some of the important issues are in these areas and point out important directions for future research efforts.

Data word indicator in a system for assembling transport data packets

Abstract: In a digital television signal processing system (10, 12 and 14-18), a special codeword, a Packet Alignement Flag (PAF), is inserted into an MPEG codeword bitstream to signify the presence of a Group of Pictures (GOP). The PAF immediately precedes a Picture Stard codeword for an "I" frame, which initiates a GOP. A data packet under construction when a PAF appears is terminated since a GOP is intended to begin at a packet boundary. Such termination may result in an abbreviated packet of less than a prescribed number of codewords needed to complete a data packet. The last word of each packet is designated as such to facilitate the subsequent combining of data packets with respective headers. An incomplete data packet is filled with null (zeroed bits) words to make up a complete data packet with a prescribed number or words.

Photonic fast packet switching

Abstract: Several approaches to photonic fast packet switching systems are presented. The wavelength-, time-, code-, and space-division approaches, including free-space photonic fast packet switching, are discussed. These approaches to photonic fast packet switching systems show that the research in this area is still in its infancy. Among various solutions, those based on a wavelength-division transport network and an electronic controller are most mature. >

Voice synchronization in packet switching networks

Abstract: An algorithm for voice synchronization for packet switching networks is presented. The algorithm has been tested both in simulation and on a real network. The algorithm runs on the TRAME packet switching network for both the Vocoder and CELP DoD voice coding standards. Some results of these tests are presented. Some details of the algorithm development and implementation are given as well. >