Showing papers on "Transmission delay published in 1998"

Policy based quality of service

Abstract: A flexible, policy-based, mechanism for managing, monitoring, and prioritizing traffic within a network and allocating bandwidth to achieve true quality of service (QoS) is provided. According to one aspect of the present invention, a method is provided for managing bandwidth allocation in a network that employs a non-deterministic access protocol, such as an Ethernet network. A packet forwarding device receives information indicative of a set of traffic groups, such as: a MAC address, or IEEE 802.1p priority indicator or 802.1Q frame tag, if the QoS policy is based upon individual station applications; or a physical port if the QoS policy is based purely upon topology. The packet forwarding device additionally receives bandwidth parameters corresponding to the traffic groups. After receiving a packet associated with one of the traffic groups on a first port, the packet forwarding device schedules the packet for transmission from a second port based upon bandwidth parameters corresponding to the traffic group with which the packet is associated. According to another aspect of the present invention, a method is provided for managing bandwidth allocation in a packet forwarding device. The packet forwarding device receives information indicative of a set of traffic groups. The packet forwarding device additionally receives information defining a QoS policy for the traffic groups. After a packet is received by the packet forwarding device, a traffic group with which the packet is associated is identified. Subsequently, rather than relying on an end-to-end signaling protocol for scheduling, the packet is scheduled for transmission based upon the QoS policy for the identified traffic group.

Method and apparatus for high-rate packet data transmission

Abstract: In a data communication system capable of variable rate transmission, high rate packet data transmission improves utilization of the forward link and decreases the transmission delay. Data transmission on the forward link is time multiplexed and the base station transmits at the highest data rate supported by the forward link at each time slot to one mobile station. The data rate is determined by the largest C/I measurement of the forward link signals as measured at the mobile station. Upon determination of a data packet received in error, the mobile station transmits a NACK message back to the base station. The NACK message results in retransmission of the data packet received in error. The data packets can be transmitted out of sequence by the use of sequence number to identify each data unit within the data packets.

Buffering in optical packet switches

Abstract: This paper consists of a categorization of optical buffering strategies for optical packet switches, and a comparison of the performance of these strategies both with respect to packet loss/delay and bit error rate (BER) performance. Issues surrounding optical buffer implementation are discussed, and representative architectures are introduced under different categories. Conclusions are drawn about packet loss and BER performance, and about the characteristics an architecture should have to be practical. It is shown that there is a strong case for the use of optical regeneration for successful cascading of these architectures.

Transparent optical packet switching: network architecture and demonstrators in the KEOPS project

Abstract: This paper reviews the work carried out in the ACTS KEOPS (Keys to Optical Packet Switching) project, describing the results obtained to date. The main objective of the project is the definition, development, and assessment of optical packet switching and routing networks, capable of providing transparency to the payload bit rate, using optical packets of fixed duration and low bit rate headers in order to enable easier processing at the network/node interfaces. The feasibility of the KEOPS concept is assessed by modeling, laboratory experiments, and testbed implementation of optical packet switching nodes and network/node interfacing blocks, including a fully equipped demonstrator. The demonstration relies on advanced optoelectronic components, developed within the project, which are described.

Method and apparatus for forward link rate scheduling

Abstract: In a communication system capable of variable rate transmission, scheduling of high speed data transmission improves utilization of the forward link and decreases the transmission delay in data communication. Each remote station is assigned one primary code channel for the duration of the communication with a cell. Secondary code channels of various types and transmission capabilities can be assigned by a channel scheduler for scheduled transmission of data traffic at high rates. Secondary code channels are assigned in accordance with a set of system goals, a list of parameters, and collected information on the status of the communication network. Secondary code channels can be grouped into sets of secondary code channels. Data is partitioned in data frames and transmitted over the primary and secondary code channels which have been assigned to the scheduled user.

Packet audio playout delay adjustment: performance bounds and algorithms

Abstract: In packet audio applications, packets are buffered at a receiving site and their playout delayed in order to com- pensate for variable network delays. In this paper, we con- sider the problem of adaptively adjusting the playout delay in order to keep this delay as small as possible, while at the same time avoiding excessive "loss" due to the arrival of packets at the receiver after their playout time has al- ready passed. The contributions of this paper are twofold. First, given a trace of packet audio receptions at a receiver, we present efficient algorithms for computing a bound on the achievable performance of any playout delay adjustment algorithm. More precisely, we compute upper and lower bounds (which are shown to be tight for the range of loss and delay values of interest) on the optimum (minimum) average playout delay for a given number of packet losses (due to late arrivals) at the receiver for that trace. Second, we present a new adaptive delay adjustment algorithm that tracks the network delay of recently received packets and efficiently maintains delay percentile information. This in- formation, together with a "delay spike" detection algorithm based on (but extending) our earlier work, is used to dy- namically adjust talkspurt playout delay. We show that this algorithm outperforms existing delay adjustment algorithms over a number of measured audio delay traces and performs close to the theoretical optimum over a range of parameter values of interest.

A system and method for a quality of service in a multi-layer network element

Abstract: A multi-layer network element (12) for forwarding received packets from an input port to one or more output ports (38) with quality of service. When output queues (54) exceed or meet a threshold value below the queue's capacity packets are randomly discarded. When the queue becomes full, the network element determines which flow caused the queue to overflow. The priority of that flow is lowered. In a multicast packet, the packet may have different priorities at each output port. Scheduling of multiple output queues at each output port uses a weight round robin approach that allocates a weight portion of packets to transmit at each time interval. A packet is not interrupted during its transmission, even if the weight portion is met during a packet's transmission. The excess number of bytes transmitted as a result of not interrupting the packet are accounted for in the next round.

An optimal topology-transparent scheduling method in multihop packet radio networks

Abstract: Many transmission scheduling algorithms have been proposed to maximize the spatial reuse and minimize the time-division multiple-access (TDMA) frame length in multihop packet radio networks. Almost all existing algorithms assume exact network topology information and do not adapt to different traffic requirements. Chlamtac and Farago (1994) proposed a topology-transparent algorithm. Following their approach, but with a different design strategy, we propose another algorithm which is optimal in that it maximizes the minimum throughput. We compare our algorithm with that of Chlamtac and Farago's and with the TDMA algorithm, and find that it gives better performance in terms of minimum throughput and minimum and maximum delay times. Our algorithm requires estimated values of the number of nodes and the maximum nodal degree in the network. However, we show that the performance of our algorithm is insensitive to these design parameters.

Method for scheduling packet data transmission

Abstract: In a packet data transmission and reception system a media access control (MAC) message is broadcast by a base station to a plurality of mobile stations. The MAC message contains packet data transmission scheduling information which allows the base station to preemptively control mobile station access to traffic channels in order to maximize the efficiency of packet data transmissions and allow scheduling consideration including priority access, quality of service and maximum bytes per transfer. The MAC message consists of a control frame structure which comprises scheduling parameters including MAC IDs fields, activity fields, and a field representing the number of free traffic channels in a cell. These parameters enable multiple mobile stations to share, in a time multiplexed fashion, traffic channels for packet data transmission on CDMA based mobile communication systems.

Clock synchronization and dynamic jitter management for voice over IP and real-time data

Abstract: A real-time receiver and method for receiving and playing out real-time packetized data are disclosed. The receiver includes a packet transmission fixed delay estimator and a packet transmission variable delay estimator. The fixed delay estimator determines, using packets received up to the current point in a conference, the non-variable portion of observed delays. This non-variable portion is subtracted from each packet's observed delay to obtain a variable delay estimate for that packet. Since variable delays actually drive the buffering time needed at the receiver to achieve smooth playout, the packet variable delay estimates can be used directly to adjust playout delay. Adaptive playout delay is preferably set aggressively low, based on observed packet variable delay estimates, to reduce data latency. Playout delay can be adjusted rapidly upwards when higher packet delays are observed, allowing rapid adaptation to network statistical variations and reducing the frequency of late packets.

A wireless fair service algorithm for packet cellular networks

Abstract: Wireless Fair Service Algorithm For Packet Cellular Networks Sonwu Lu Thyagarajan Nandagopal Vaduvur Bharghavan Coordinated Science Laboratory University of Illinois email: {sIu, thyagu, bharghav}~timely. crhc .uiuc. edu b order to support diverse communication-intensive redtirne and non red-time data flows over a scarce, varying and shared wireless channel with location-dependent and bursty errors, we defie a service model that has the following charxteristi~ short-tern fairness among flows which perceive a clean channel, worst-case delay bounds for packets, short-term throughput bounds for flows with clean channels and Iong-tem throughput bounds for dl flows with bounded channel error, optimal schedulable region, and support for both delay sensitive and error sensitive data flows. We present a wireless fair service algorithm, and show that it achieves the requirements of the service model through both analysis and simulation. The key aspects of the dg~ rithm are the following (a) an enhanced fair queueing based service scheme that supports decoupling of delay and bandwidth, (b) graceful service compensation for lagging flows and graceful service degradation for leading flows, (c) support for red-time delay sensitive flows as well as non realtime error sensitive flows, and (d) implementation of the wireless fair service rdgorithm within the fiarnework of the simple and robust CSMA/CA wireless medium access pro tocol.

Mechanism for dispatching packets via a telecommunications network

Abstract: A mechanism for dispatching a sequence of packets via a telecommunications network includes a queue for packets for transmission and a queue controller responsive to receipt of a new packet for transmission to compare parameters of the new packet to parameters of any packet already in the queue, the queue controller determining whether to queue or drop the new packet depending on the result of the comparison(s). The queue can be implemented as a linked list of packet entries with individual pointers to the respective packets concerned. The queue entries can include details relating to the packet including data relating to the information flow and also the packet identity. In a TCP environment, the flow information can include the source IP address and the source TCP port, as well as the destination IP address and the destination TCP port. The identity information can include sequence numbers and acknowledgement numbers for the packet concerned. In order to optimize network usage, it can be useful to drop some packets at a routing node. A decision to drop a packet can be made if the new packet and a queued packet relate to the same information flow, the new packet sequence number equals the queued packet sequence number and the new packet acknowledgement number is less than the queued packet acknowledgement number. The new packet is dropped where the new packet is a retransmission of a queued packet and the length of the queued packet is greater than or equal to that of the new packet. A queued packet is replaced by a new packet when the new packet is determined to be a retransmission of the queued packet and the length of the new packet is greater than that of the queued packet.

Packet switching method with time-based routing

Abstract: This invention describes a method for transmitting and forwarding packets over a switching network using time information. The network switches maintain a common time reference, which is obtained either from an external source (such as GPS—Global Positioning System) or is generated and distributed internally. The time intervals are arranged in simple periodicity and complex periodicity (like seconds and minutes of a clock). A data packet that arrives to an input port is switched to an output port based on its order or time position in the time interval in which it arrives at the switch. The time interval duration can be longer than the time duration required for transmitting a data packet, in which case the exact position of a data packet in its forwarding time interval is predetermined. This invention provides congestion-free data packet switching for data packets for which capacity in their corresponding forwarding links and time intervals is reserved in advance. Furthermore, such data packets reach their destination, which can be one or more (i.e., multicast) in predefined time intervals, which guarantees that the delay jitter is smaller than or equal to one time interval

Wavelength routing-based photonic packet buffers and their applications in photonic packet switching systems

Abstract: Photonic packet buffers are essential components in photonic packet switching systems. We present a wavelength routing-based photonic packet buffer based on a state-of-the-art arrayed-waveguide grating (AWG) multiplexer. We show how this new packet buffer can be effectively used in the implementation of photonic packet switching systems. We also propose and examine two different photonic packet switch architectures.

Method and apparatus for time efficient retransmission using symbol accumulation

Abstract: An efficient retransmission of data using symbol accumulation wherein the packet received in error is retransmitted at a lower energy-per-bit level concurrently in the same frame with the new packet. The destination device receives the data transmission and retransmission, demodulate the signal, and separates the received data into the new and retransmitted packet. The destination device then accumulates the energy of the retransmitted packet with the energy already accumulated for the packet received in error and decodes the accumulated packet. The accumulation of the additional energy provided by the subsequent retransmissions improves the probability of a correct decoding. The throughput rate can be improved since the packet received in error is retransmitted concurrently with the transmission of the new data packet. The capacity is maximized since the retransmission of the packet received in error is at a lower energy level than that of the new packet.

Method for pacing data flow in a packet-based network

Abstract: A method for pacing data flows in packet switched networks by arranging data transmission over a period of time based upon a set of ascertainable factors about the underlying transmission link to derive an intersegment transmission interval. The intersegment transmission interval can be used to pace either data packets or acknowledgment packets. The method is especially useful for pacing the transmission of data in a digital data packet communication environment having a plurality of digital packet transmission stations inter-connectable in a data path and employing the Transmission Control Protocol (TCP) suite.

Method for setting time stamp in syt field of packet headers for ieee-1394 devices

Abstract: Isochronous data packets transmitted within a digital network having a bus architecture that complies with the IEEE-1394 Standard for a High Performance Serial Bus are stamped with a presentation time stamp value determined according to a computed packet rate for the data For the case where the presentation time stamp field of a first packet of a second frame of data for transmission in the digital network is set with the presentation time value, the packet rate may be computed by measuring a difference between a desired presentation time value of a first packet in a first frame of the data and an actual transmission time of the first packet of the first frame The first frame preceding the second frame in time of transmission within the network

Implementing distributed packet fair queueing in a scalable switch architecture

Abstract: To support the Internet's growth, there is a need for cost effective switching technologies that can simultaneously provide high capacity switching and advanced QoS. Unfortunately, these two goals are largely believed to be contradictory in nature. To support QoS, sophisticated packet scheduling algorithms, such as fair queueing, are needed to manage queueing points. However, the bulk of current research in packet scheduling algorithms assumes an output buffered switch architecture, whereas most high performance switches are input buffered. While output buffered systems may have the desired QoS, they lack the necessary scalability. Input buffered systems, while scalable, lack the necessary QoS features. We propose the construction of switching systems that are both input and output buffered with the scalability of input buffered switches and the robust QoS of output buffered switches. We call the resulting architecture distributed packet fair queueing (D-PFQ) as it enables physically dispersed line cards to provide a service that closely approximates an output-buffered switch with fair queueing. By equalizing the growth of the virtual time functions across the switch system, most of the PFQ algorithms in the literature can be properly defined for distributed operation. We present our system using a cross bar for the switch core. Buffering techniques are used to enhance the system's latency tolerance, which enables the use of pipelining and variable packet sizes internally. We evaluate the delay and bandwidth sharing properties.

Packet switching with common time reference over links with dynamically varying delays

Abstract: The invention describes a method for transmitting and forwarding packets over a packet switching network wherein the delay between two switches increases, decreases, or changes arbitrarily over time. Packets are being forwarded over each link inside the network in predefined periodic time intervals. The switches of the network maintain a common time reference, which is obtained either from an external source (such as GPS--Global Positioning System) or is generated and distributed internally. The time intervals are arranged with simple periodicity and complex periodicity (like seconds and minutes of a clock). When the delay increases at some point of time, a packet may be late for its predefined forwarding time interval. In such case, the packet is delayed until the next time interval of its virtual pipe. When the link delay decreases, packets are buffered until the first time interval of its virtual pipe.

Codec-independent technique for modulating bandwidth in packet network

Abstract: The size of packet payloads are varied according to the amount of congestion in a packet network. More data is put in packet payloads when more congestion exits in the packet network. When network congestion is high, less network bandwidth is available for transmitting packets. Accordingly, the packet payloads are transmitted with larger payloads to reduce the percentage of overhead in each packet. When there is little or no network congestion smaller packet payloads are transmitted. The additional overhead created in transmitting smaller packets is acceptable when there is little or no network congestion because the network currently has excess bandwidth. Thus, the packet payloads are dynamically adjusted to use network resources more effectively.

Method of controlling communication resources

Abstract: According to the method of the invention, connections are divided into at least two different connection classes according to their requirements for transmission delay. The control system of the base station subsystem maintains a record of the transmission needs of the users logged in different categories and based thereon divides the available radio resources into slots of suitable capacity. For connections with stringent requirements for transmission delay, circuit-switched connections are allocated with a bandwidth which can be controlled dynamically. Then from the resource pool still unassigned after the resource allocation to the circuit-switched connections, a sufficient amount of resources are allocated on a time-limited basis allocation for each allocation period to connections having a higher tolerance for delay so as to accomplish transmission, e.g. of a given amount of data.

System for providing stream based and packet based services

Abstract: This invention proposes a novel common packet data channel (CPDC) system structure and method to efficiently provide variable-rate packet data services in CDMA system. This system uses common code channels to serve packet data calls for a plurality of packet data users within a cell or sector. On the forward link (base station-to-mobile), an ATM-type multiplexing scheme is employed on a common code channel, while a spread ALOHA-type random access scheme is used on the reverse link (mobile-to-base station). The overhead due to call setup and channel assignment in a conventional circuit-switched system is not required in this system. The delay due to the call setup is reduced. In addition, ATM cells can be directly transmitted over the air interface. Novel transceiver architectures for the terminal and base station for implementing a system as well as particular channel signal structures are disclosed.

Route selection for path balancing in connection-oriented packet switching networks

Abstract: In a methodology for route selection for path balancing in a connection- oriented packet switching network, a comparison is made of the measures of utilizations of a network resource by at least two links of a link group in the network, and at least one link is identified as a candidate for carrying traffic path based on the comparison. Network traffic path on the link is then moved to the candidate link to reduce variation in the utilization of the network resource by each of the links.

Method for transmitting packet switched data in a mobile communications system

Abstract: The invention relates to a method for transmitting packet switched data in a mobile communications system using an ARQ protocol. In the method the receiver orders the desired transmission units. The transmitter sends transmission units according to the order. Ordering and transmission are repeated until the quality measured by the receiver from each packet exceeds the predetermined quality value and the data which is to be transmitted and consists of packets is transmitted from the transmitter to the receiver. The invention also relates to a mobile communications system implementing the method of the invention.

Dynamically configurable buffer for a computer network

Abstract: A dynamically configurable network buffer includes a buffer manager organizing a buffer memory into a set of uniform sized packet buffers, each of which is large enough to store the largest possible data packet that may be transmitted through the network switch. The buffer manager further subdivides each packet buffer into a set of smaller packet cells of uniform size. When an incoming data packet arrives at the network buffer, the buffer manager determines its size. If the packet is too large to be stored in a single packet cell, the buffer manager stores the packet by itself in an unoccupied packet buffer. If the packet is small enough to be stored in a single packet cell, the buffer manager stores the packet in an unoccupied packet cell. The network buffer can increase or decrease packet cell size in response to input configuration data.

System for user-space network packet modification

Abstract: A system for user-space packet modification, including a set of kernel code and a user-level application programming interface (API). The system facilitates creation of a special socket for passing packets between kernel space and user space. The system in turn facilitates creation and application of a packet filter associated with the socket, in order to trap incoming or outgoing packets being processed in the kernel at a designated point in a protocol stack. Once a packet is trapped, it is moved through the socket into user space, thereby at least temporarily preventing the protocol stack from further processing the packet. In user space, an application may operate on the packet, for instance, modifying aspects of the packet or deleting the packet altogether. The system in turn facilitates injection of a packet from user space into kernel space, and into a designated point in the protocol stack for desired stack processing.

Method for assigning optimal packet lengths in a variable rate communication system

Abstract: A method for assigning optimal packet lengths in a variable rate communication system capable of data transmission at one of a plurality of data rates. The packet lengths for the data rates are selected such that the maximum throughput rate is achieved while conforming to a fairness criteria. The fairness criteria can be achieved by restricting the packet length assigned to each data rate to a range of value, or Limin ≤ L?i? ≤ Li?max?. The packet lengths for all data rates are first initialized to the maximum packet lengths for those data rates. Then, for each date rate, a determination is made whether another packet length assignment would result in improved throughput rate. If the answer is yes, the packet length for this data rate is reassigned and the throughput rate with the updates packet length assignments is recomputed. The process is repeated for each data rate until all data rates have been considered. The throughput rate can be calculated using a probabilistic model or a deterministic model.4

Software interface between switching module and operating system of a data packet switching and load balancing system

Abstract: A data packet switching system comprises a plurality of network interfaces each adapted to be coupled to respective external networks for receiving and sending data packets to and from the external networks via a particular communication protocol. The data packet switching system further includes a plurality of symmetrical processors, including a first processor providing a control processor and remaining ones of the processors each providing data packet switching processors. The data packet switching processors are coupled to the plurality of network interfaces. The control processor further includes a user portion and an operating system portion. The operating system portion is provided with a pseudo-network driver that appears to be a network interface to user application programs operating on the user portion of the control processor. A memory space is shared by the control processor and the data packet switching processors. The data packet switching processors route an incoming data packet directed to a user application program to the memory space. The pseudo-network driver retrieves the incoming data packet from the shared memory space and provides the data packet to the user application program.

Method and apparatus for selectively discarding packets

Abstract: In a packet voice system, discarding of a packet is performed as a function of previously discarded packets. In one embodiment, a packet voice system includes an ATM Adaptation Layer Type 2 (AAL2) and Service Specific Convergence Sublayer (SSCS) System. In this system, a transmission buffer stores AAL2 voice packets for transmission, each AAL2 voice packet comprising a sequence number, the values of which range from 0 to n−1, and a source identifier, k. When traffic congestion is detected, the transmitter portion of the SSCS System selectively discards one packet from a source k at the output of the transmission buffer if no packet from source k was dropped in either the last n−1 packets or over a predefined prior interval of time. Another embodiment of the invention discards packets at the input of the transmission buffer.

Packet switching fabric using the segmented ring with resource reservation control

Abstract: A packet switching fabric includes a data ring, a control ring, a plurality of network links each coupled to at least one network node, and a plurality of switching devices coupled together by the data ring and the control ring so that the network links can be selectively communicatively coupled. Each of the switching devices includes: a data ring sub-system for transmitting and receiving bursts of data via data ring channels concurrently active on the data ring; a network interface coupled to the data ring sub-system and having at least one network port for transmitting and receiving data packets to and from one of the network links, the network interface also having a packet buffer for storing the data packets, the packet buffer providing bursts of packet data to the data ring sub-system via a plurality of concurrently active packet buffer channels; and a control ring sub-system coupled to the data ring sub-system and to the network interface and being responsive to control messages received from an adjacent one of the devices via the control ring, and operative to develop and transmit the control messages to an adjacent one of the devices via the control ring, the control messages for reserving bandwidth resources used in setting up and controlling the data ring channels and the packet buffer channels, the control ring sub-system also being operative to perform queuing operations for controlling the transfer of the bursts of packet data from the packet buffer to the data ring sub-system.