Showing papers on "Transmission delay published in 1997"

End-to-end Internet packet dynamics

Abstract: We discuss findings from a large-scale study of Internet packet dynamics conducted by tracing 20,000 TCP bulk transfers between 35 Internet sites. Because we traced each 100 Kbyte transfer at both the sender and the receiver, the measurements allow us to distinguish between the end-to-end behaviors due to the different directions of the Internet paths, which often exhibit asymmetries. We characterize the prevalence of unusual network events such as out-of-order delivery and packet corruption; discuss a robust receiver-based algorithm for estimating "bottleneck bandwidth" that addresses deficiencies discovered in techniques based on "packet pair"; investigate patterns of packet loss, finding that loss events are not well-modeled as independent and, furthermore, that the distribution of the duration of loss events exhibits infinite variance; and analyze variations in packet transit delays as indicators of congestion periods, finding that congestion periods also span a wide range of time scales.

Method for explicit data rate control in a packet communication environment without data rate supervision

Abstract: A method for explicit data rate control is introduced into a packet communication environment (10) which does not have data rate supervision by adding latency to the acknowledgement (ACK) packet and by adjusting the size of the flow control window associated with the packet in order to directly control the data rate of the source data at the station (12 or 14) originating the packet.

System and method for equalizing the delay time for transmission paths in a distributed antenna network

Abstract: A communication system and method are provided for equalizing delay of transmission paths in a distributed antenna network. The distributed antenna network includes a plurality of remote antenna units, a central unit or a base station connected to the remote antenna units by transmission media, where each connection between the base station and one of the remote antenna units forms a separate transmission path having an associated delay time, a delay detector for determining the associated delay time of the separate transmission paths for each of the remote antenna units, and a delay compensator for adjusting the associated delay times in response to the delay detectors so that all of the associated delay times are substantially equalized. The system and method allow the delay parameters for the entire network to be set upon installation and then to be periodically updated without physical intervention by an operator. The detection and compensation allow for equalization of delay time differences that could not otherwise be compensated in the base stations of mobile stations of conventional systems and methods. Furthermore, the equalization synchronizes the bursts so that air frame timing between cells served by the remote antenna units is enhanced and the hand-off performance therebetween is improved.

Error correction system for packet switching networks

Abstract: A system for correcting errors in the transmission of data packets between a source and a receiver. The source sends data packets to the client unit and server unit. The system uses the client unit and the server unit to send a repaired packet stream to a receiver when an error is detected. The client unit detects errors in the packet stream and sends retransmission requests of the lost data packets to the server unit. The server unit retransmits the lost data packet to the client unit, which then corrects the packet stream by inserting the lost packet into the proper time order and transmitting the repaired packet stream to the receiver.

Improving performance of TCP over wireless networks

Abstract: Transmission Control Protocol (TCP) assumes a relatively reliable underlying network where most packet losses are due to congestion. In a wireless network, however, packet losses will occur more often due to unreliable wireless links than due to congestion. When using TCP over wireless links, each packet loss on the wireless link results in congestion control measures being invoked at the source. This causes severe performance degradation. In this paper, we study the effect of: burst errors on wireless links; packet size variation on the wired network; local error recovery by the base station; and explicit feedback by the base station, on the performance of TCP over wireless networks. It is shown that the performance of TCP is sensitive to the packet size, and that significant performance improvements are obtained if a good packet size is used. While local recovery by the base station using link-level retransmissions is found to improve performance, timeouts can still occur at the source, causing redundant packet retransmissions. We propose an explicit feedback mechanism, to prevent these timeouts during local recovery. Results indicate significant performance improvements when explicit feedback from the base station is used. A major advantage of our approaches over existing proposals is that no state maintenance is required at any intermediate host. Experiments are performed using the Network Simulator (NS) from Lawrence Berkeley Labs. The simulator has been extended to incorporate wireless link characteristics.

Optimal multiplexing on a single link: delay and buffer requirements

Abstract: This paper is motivated by the need to provide per-session quality of service guarantees in fast packet-switched networks. We address the problem of characterizing and designing scheduling policies that are optimal in the sense of minimizing buffer and/or delay requirements under the assumption of commonly accepted traffic constraints. We investigate buffer requirements under three typical memory allocation mechanisms which represent tradeoffs between efficiency and complexity. For traffic with delay constraints we provide policies that are optimal in the sense of satisfying the constraints if they are satisfiable by any policy. We also investigate the tradeoff between delay and buffer optimality, and design policies that are "good" (optimal or close to) for both. Finally, we extend our results to the case of "soft" delay constraints and address the issue of designing policies that satisfy such constraints in a fair manner. Given our focus on packet switching, we mainly concern ourselves with nonpreemptive policies, but one class of nonpreemptive policies which we consider is based on tracking preemptive policies. This class is introduced and may be of interest in other applications as well.

Selective retransmission for efficient and reliable streaming of multimedia packets in a computer network

Abstract: An efficient and reliable transmission protocol for transmitting multimedia streams from a server to a client computer over a diverse computer network including local area networks (LANs) and wide area networks (WANs) such as the internet. The client computer includes a playout buffer for temporary storage of incoming data packets. When the client computer detects that a data packet has not arrived at said client computer by an expected time of arrival (ETA), a round trip time for the data packet is computed. The round trip time is an estimate of a period beginning from the time a retransmission request is sent to from the client computer to the stream server till the time a copy of the missing data packet is received at the client computer from the stream server in response to the retransmission request. If the round trip time is less than the time remaining before the missing packet is no longer useful to the on-demand application, then a retransmission request packet is sent to the server. Conversely if the round trip time is greater than the time remaining, i.e., the missing packet is likely to arrive after the usefulness of the packet has expired, then sending a retransmission request is likely to result in the late arrival of the missing data packet. Accordingly, the missing packet is discarded. This selective retransmission protocol can also be practiced with dynamic bandwidth selection wherein the transmission rate is dynamically matched to the available bandwidth capacity of the network connection between the server and the client computer.

Apparatus, system and method for a unified circuit switched and packet-based communications system architecture with network interworking functionality

Abstract: An apparatus (105), system (100) and method is provided for a unified circuit switched and packet-based communications system architecture having network interworking functionality. The apparatus (105) includes a circuit switched network interface (130), with the circuit switched network interface coupleable to a network switch (135) for communication of a circuit switched network protocol signal with a circuit switched network (160); a packet-based network interface (140), the packet-based network interface coupleable to a router (115) for communication of a packet-based network protocol signal with a packet-based network (150); a transceiver (120) coupleable to the communications channel (103) for the transmission and reception of a first protocol signal; and a communications controller (145), the communications controller coupled to the circuit switched network interface, to the packet-based network interface and to the transceiver, the communications controller responsive, through a set of program instructions and an interworking function, to interconvert the first protocol signal with the circuit switched network protocol signal and to interconvert the first protocol signal with the packet-based network protocol signal.

Distributed VLAN mechanism for packet field replacement in a multi-layered switched network element using a control field/signal for indicating modification of a packet with a database search engine

Abstract: A system and method for updating packet headers using hardware that maintains the high performance of the network element. In one embodiment, the system includes an input port process (IPP) that buffers the input packet received and forwards header information to the search engine. The search engine searches a database maintained on the switch element to determine the type of the packet. In one embodiment, the type may indicate whether the packet can be routed in hardware. In another embodiment, the type may indicate whether the packet supports VLANs. The search engine sends the packet type information to the IPP along with the destination address (DA) to be updated if the packet is to be routed, or a VLAN tag if the packet has been identified to be forwarded to a particular VLAN. The IPP, during transmission of the packet to a packet memory selectively replaces the corresponding fields, e.g., DA field or VLAN tag field; the modified packet is stored in the packet memory. Associated with the packet memory are control fields containing control field information conveyed to the packet memory by the IPP. An output port process (OPP) reads the modified input packet and the control field information and selectively performs additional modifications to the modified input packet and issue control signals to the output interface (i.e., MAC). The MAC, based upon the control signals, replaces the source address field with the address of the MAC and generates a CRC that is appended to the end of the packet.

Time-spread multiple-access (TSMA) protocols for multihop mobile radio networks

Abstract: This paper introduces a novel technique called protocol threading, yielding a deterministic protocol that gives a guaranteed upper bound on the transmission delay of each packet at every node in a multihop mobile network. By eliminating the maximum degree constraint, the new method improves upon existing time-spread multiple-access (TSMA)-type protocols while preserving the advantages of the deterministic operation and topology transparency. We introduce the protocol threading solution, derive the maximum delay bound in a mobile topology, and analyze the performance of the protocol.

Method for constructing adaptive packet lengths in a congested network

Abstract: In a network, the efficiency of data transfer is improved. The network includes packets of data (100, 120). The invention begins by preserving an original packet (100). A larger packet (140) is then constructed by combining two packets (100, 120). Either the original packet (100) or the larger packet (140) is transmitted over the network. The original packet (100) is transmitted if the media/port (52) becomes available for transmission before the larger packet (140) is constructed, and the larger packet (140) is transmitted if the constructing of the larger packet (140) is completed before the media (52) becomes available. In a network with cells, a packet is constructed by combining cells. This packet is built until the media becomes available for transmission. The size and composition of this packet is independent of an original packet's size and composition.

Network and system concepts for optical packet switching

Abstract: An overview of the characteristics and challenges of optical packet switching is given, illustrating its potential advantages within future nodes and networks, describing basic system functionalities. The opportunities introduced by the ACTS KEOPS project on all-optical packet-switching networks are highlighted, based partially on the outcome of the RACE ATMOS project, which is also considered in this article.

Packet scheduling scheme for improving short time fairness characteristic in weighted fair queueing

Abstract: A packet scheduling scheme which is capable of improving the fairness characteristic in a short time scale by suppressing the burstiness of traffic compared with the conventional weight fair queueing algorithm such as DRR. Packets are held in a plurality of packet queues by inputting arrived packets into the packet queues. Then, an output packet queue is sequentially selected from the packet queues, according to a prescribed criterion based on an amount of packets currently transmittable by each packet queue, such that the output packet queue is selected to be different from a previously selected output packet queue when there are more than one packet queues that satisfy the prescribed criterion. Then, a top packet is outputted from the sequentially selected output packet queue.

Packet processing device and mobile computer with reduced packet processing overhead

Abstract: A packet processing and packet transfer scheme capable of reducing the packet processing overhead by eliminating a need to decrypt and re-encrypt the entire packet at a time of relaying encrypted packets In a packet processing device for relaying encrypted packets, a packet transferred to the packet processing device is received, where the packet has a packet processing key to be used in a prescribed packet processing with respect to a data portion of the packet, and the packet processing key is encrypted by using a first master key shared between a last device that applied a cipher communication related processing to the packet and the packet processing device Then, the packet processing key in the received packet is decrypted, without carrying out the prescribed packet processing with respect to the data portion of the packet, and the decrypted packet processing key is re-encrypted by using a second master key shared between a next device to apply the cipher communication related processing to the packet and the packet processing device Then, the packet with the re-encrypted packet processing key encoded therein is transmitted toward a destination of the received packet

Dynamic global packet routing in wireless networks

Abstract: We consider schemes for reuse-efficient packet access in wireless data networks. We show that computing the maximum ergodic packet arrival rate is NP-hard. We give an upper bound on the maximum ergodic throughput in terms of the eigenvalues of matrices related to the path-gain matrix. We present simple, practical heuristic algorithms which exhibit good throughput and packet delay and report on results of preliminary simulations. More sophisticated algorithms that yield optimal throughput are also presented. A recent result of McKeown, Anantharam and Walrand (1996) on scheduling of input-queued switches is obtained as a by-product.

Communications system having distributed control and real-time bandwidth management

Abstract: A communications system and method utilizes peer-to-peer architecture to provide "quality" real-time communications in a non-guaranteed, packet-based network. Quality real-time communications in a non-guaranteed, packet-based network is achieved by (i) minimizing packet loss, (ii) minimizing latency, and (iii) minimizing the artifacts caused by packet loss. The system distributes control of functionality to the peer communication devices, thereby eliminating the need for a central processor to establish peer-to-peer communications. Each of the peer communication devices has the capability to respond to variations in network loading to provide quality real-time communications.

Router device and network system using the same

Abstract: A router device facilitates management of connection of a manager and assures communication quality of each of the connections. The router device determines a communication quality required by the communication of the packet in reference to a protocol information, an application acknowledging information and a priority information, transmits the control packet having the determined communication quality set therein to another router device contained in the same connection and further performs a traffic control (a preferential processing of a packet or the like) satisfying a communication quality indicated by the transmitted or received control packet in respect to the packet of the aforesaid connection.

Traffic scheduling method, system and article of manufacture for a wireless access to an asynchronous transfer mode network

Abstract: A method, system and article of manufacture for exchanging data between an asynchronous network and a synchronous network is presented. Data is exchanged between the networks in a sequence of time frames which are partitioned into downlink, uplink and contention periods. These downlink, uplink and contention periods are further divided into time slots, each of which carries either data or control cells between the networks. Cells of data are allocated to the time slots according to their cell deadlines which are proportional to the transmission delay of the network connection over which the cells are to be carried. This cell allocation is then stored in a slot map and communicated to all nodes within the networks in order to facilitate the data transmission.

Asynchronous packet switching

Abstract: A multiprocessor system includes a plurality of nodes and an interconnect that includes routers. Each node includes a reliable packet mover and a fast frame mover. The reliable packet mover provides packets to the fast frame mover which adds routing information to the packet to form a frame. The route to each node is predetermined. The frame is provided to the routers which delete the route from the routing information. If the frame is lost while being routed, the router discards the frame. If the packet is received at a destination node, the reliable packet mover in that node sends an acknowledgement to the source node if the packet passes an error detection test. The reliable packet mover in the source node resends the packet if it does not receive an acknowledgement in a predetermined time. The fast frame mover randomly selects the route from a plurality of predetermined routes to the destination node according to a probability distribution.

Method and apparatus for packet transmission

Abstract: In transmitting a packet string having variable packet intervals by converting the packet string into that having even packet intervals with each of the packets being attached with a time stamp as information for reproducing original packet string, when the value of the time stamp, which is decided by adding a specified offset time to the synchronization time, is not smaller than the value of the time stamp attached to the previous packet, the time stamp is attached to the packet and the packet is transmitted. If the value of the time stamp becomes not more than the value of the time stamp attached to the previous packet as a result of providing shortened offset time due to increased bit rate of the original packet string, the packet is discarded so as to protect the transmission from being stopped.

Method and system for generating data packets on a heterogeneous network

Abstract: An improved method and system for generating packets for transmission over different routes on a network is presented. In a large network such as the Internet, each route over the network carries different size packets which are not compatible unless they are fragmented into smaller segments. Initially, the technique determines a maximum transmission unit (MTU) capable of being transmitted over a predetermined route. Next, the size of each packet to be transmitted over the network is compared with the MTU size. If the comparison indicates the packet is larger than the MTU, the packet must be processed further before it can be transmitted over the route. The additional processing initially divides the total number of transmission units contained within the packet by the MTU value. The integer result of this division is temporarily stored in a DCOUNT variable and the remainder of the division in a RCOUNT variable. If the remainder in RCOUNT is non-zero, the value in DCOUNT is incremented by one. The DCOUNT value indicates the minimum number of datagrams for sending a packet using the present technique. Next, the transmission units contained in the original packet are redistributed equally into DCOUNT packets and then sent over the network route.

Multiplexing of voice and data minicells

Abstract: Segmenting, multiplexing and transporting user data packets in a telecommunication system that employs Asynchronous Transfer Mode can be accomplished more effectively by assigning a transmission priority code to each segmented user data packet based on the type of data contained therein. Data that is highly sensitive to transmission delays (e.g., voice data) will be assigned a high priority, while data that is less sensitive to transmission delays (e.g., signal strength measurement data) will be assigned a lower priority. When the user data packet segments are assembled into segment minicells and multiplexed into the ATM cell stream, those with the highest priority will be inserted first so that they experience the least amount of transmission delay.

Single-chip architecture for shared-memory router

Abstract: The invention provides a single-chip method. The method includes a memory shared among packet buffers for receiving packets, packet buffers for transmitting packets, and packet header buffers for packet forwarding lookup. Accesses to that shared memory are multiplexed and prioritized. Packet reception is performed with relatively high priority, packet transmission is performed with medium priority, and packet forwarding lookup is performed with relatively low priority. The single-chip method includes circuits for serially receiving packet header information, converting that information into a parallel format for transmission to an SRAM for lookup, and queuing input packets for later forwarding at an output port. Similarly, the single-chip method includes circuits for queuing output packets for transmission at an output port, receiving packet forwarding information from the SRAM in a parallel format, and converting packet header information from output packets into a serial format for transmission. The single-chip method also includes a region in its shared memory for a packet forwarding table, and circuits for performing forwarding lookup responsive to packet header information.

Error correction and loss recovery of packets over a computer network

Abstract: A method of recovering a lost or corrupted data packet of a plurality of data packets transmitted by a transmitting computer system to a target computer system over an unreliable computer network. In one embodiment, the transmitting computer system generates parity information by padding the plurality of data packets to the length of the longest data packet with a suitable bit pattern. The parity information is then generated using a suitable algorithm, such as an exclusive OR (XOR) operation. The parity information and the packet lengths of the data packets are used to form the parity packet. The transmitting computer system then sends both the data packets and the parity packet to the target computer. If the target computer system detects a lost or corrupted data packet, the target computer attempts to reconstruct the lost or corrupted data packet. The received uncorrupted data packets are padded with the suitable bit pattern described above. Note that it is useful to pad up to the length of the lost data packet. The inverse operation of the parity generating algorithm is then applied to the padded data packets and the parity packet to reconstruct the lost or corrupted data packet.

Rate control techniques for efficient high speed data services

Abstract: Methods for controlling data rate allocations to data packet users transmitting packet data over a CDMA cellular communication network are defined which comprises the steps of: evaluating traffic channels and radio capacity allocated for packet data services within the network to determine an available resource for a packet data transmission; employing a rate control algorithm to determine a data rate allocation for the packet data transmission; and limiting the transmit power of a transmitter to provide the determined data rate allocation for the packet data transmission. The methods include a rate control algorithm which determines data rate allocation using a transmission power budget technique and a rate control algorithm which determines data rate allocation using a current system load technique.

Packet switch and congestion notification method

Abstract: A packet switch for setting a connection between a transmission source of a packet and a reception destination thereof so as to perform communication. The invention includes a packet buffer which includes at least one input port and a plurality of output ports. An input packet from the input port is delivered to at least one output port in accordance with address information of the input packet and connection information having been set in the packet switch at the time of setting the connection between the transmission source and the reception destination. A bandwidth management packet for giving notice of a congested state of the packet switch is transferred on the connection. The invention further includes a register which holds threshold value information for indicating an amount of use of the packet buffer that causes congestion, a counter which provides a count representative of a current amount of use of the packet buffer, a comparator which compares the count from the counter and the threshold value information from the register and outputs a result of the comparison, and a congestion decision/notification circuit which writes congestion notification information into the bandwidth management packet based on a result of the comparison by the comparator.

Packetized data transmissions in a switched router architecture

Abstract: A switched router for transmitting packetized data concurrently between a plurality of devices coupled to the switched router. The devices are coupled to the I/O ports of the switched router. The switched router is then programmed to route packets of data from various source ports to several destination ports. Different packets may be transmitted concurrently through the switched router. The packets are comprised of a command word containing information corresponding to packet routing, data format, size, and transaction identification. Furthermore, the command word may include a destination identification number for routing the packet to a destination device, a source identification number used by a destination device to send back responses, a transaction number to tag requests that require a response, and a packet type value indicating a particular type of packet. In addition, there may be bits within a packet used to indicate a coherent transaction, guarantee bandwidth, an error during transmission, or a sync barrier for write ordering. Other types of packets may include a fetch and operation packet with increment by one, a fetch and operation packet with decrement by one, a fetch and operation packet with clear, a store and operation packet with increment by one, a store and operation packet with decrement by one, a store and operation packet with a logical OR, and a store and operation packet with a logical AND.

Method and apparatus for transmitting memory requests by transmitting portions of count data in adjacent words of a packet

Abstract: A high speed bus system in which at least one master device, such as a processor and at least one DRAM slave device are coupled to the bus. An innovative packet format and device interface which utilizes a plurality of time and space saving features in order to decrease the die size of the device receiver and decrease the overall latency on the bus is provided. In the preferred embodiment the request packet is transmitted on ten multiplexed transmission lines, identified as BusCtl and BusData 8:0!. The packet is transmitted over six sequential bus cycles, wherein during each bus cycle, a different portion of the packet is transmitted. The lower order address bits are moved ahead of the higher order address bits of the memory request. This enables the receiving device to process the memory request faster as the locality of the memory reference with respect to previous references can be immediately determined and page mode accesses on the DRAM can be initiated as quickly as possible. The type of memory access is arranged over a plurality of clock cycles, placing the more critical bits first. The count of blocks of data requested is arranged to minimize the number of bit positions in the packet used and therefore the number of transmission lines of the bus and the number of bus receiver contacts on the receiving device.

Bounce diagram: a user interface for graphical exploration of packet trace information

Abstract: A user interface for a protocol analyzer or similar network management software product provides a graphical representation of the behavior of packets in a packet trace with respect to time, graphically showing a transmission time, source node and destination node. In a preferred implementation, the user interface, called a “bounce diagram,” contains a time axis, and a number of node lines parallel thereto, each node line associated with a node label for a node in the packet trace. Each packet in the packet trace is represented by a packet arrow which extends from a node line for the source node to a node line for the destination node of the packet. The packet arrow is preferably color coded to indicate the size of the packet. Preferably, there is a packet density graph which indicates in each time interval a number or percentage of packets of the packet trace transmitted during the time interval. The packet density graph may be color coded to indicate the average size of packets during each time interval. The bounce diagram is interactive and responds to a user selection of a packet arrow by displaying a pop-up window with the packet size, source and destination node addresses, transmission time, and protocol decodes for the packet. The user may also zoom in and out of the diagram by selecting an area thereof, with the diagram automatically rescaling the time axis to correspond to the time within the selected area.

Power Consumption in Packet Radio Networks (Extended Abstract)

Abstract: In this paper we study the problem of assigning transmission ranges to the nodes of a multi-hop packet radio network so as to minimize the total power consumed under the constraint that adequate power is provided to the nodes to ensure that the network is strongly connected (i.e., each node can communicate along some path in the network to every other node). Such assignment of transmission ranges is called complete. We also consider the problem of achieving strongly connected bounded diameter networks.