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Showing papers on "Packet loss published in 2000"


Journal ArticleDOI
28 Aug 2000
TL;DR: A mechanism for equation-based congestion control for unicast traffic that refrains from reducing the sending rate in half in response to a single packet drop, and uses both simulations and experiments over the Internet to explore performance.
Abstract: This paper proposes a mechanism for equation-based congestion control for unicast traffic. Most best-effort traffic in the current Internet is well-served by the dominant transport protocol, TCP. However, traffic such as best-effort unicast streaming multimedia could find use for a TCP-friendly congestion control mechanism that refrains from reducing the sending rate in half in response to a single packet drop. With our mechanism, the sender explicitly adjusts its sending rate as a function of the measured rate of loss events, where a loss event consists of one or more packets dropped within a single round-trip time. We use both simulations and experiments over the Internet to explore performance.

1,458 citations


Proceedings ArticleDOI
14 May 2000
TL;DR: This work proposes two efficient schemes, TESLA and EMSS, for secure lossy multicast streams, and offers sender authentication, strong loss robustness, high scalability and minimal overhead at the cost of loose initial time synchronization and slightly delayed authentication.
Abstract: Multicast stream authentication and signing is an important and challenging problem. Applications include the continuous authentication of radio and TV Internet broadcasts, and authenticated data distribution by satellite. The main challenges are fourfold. First, authenticity must be guaranteed even when only the sender of the data is trusted. Second, the scheme needs to scale to potentially millions of receivers. Third, streamed media distribution can have high packet loss. Finally the system needs to be efficient to support fast packet rates. We propose two efficient schemes, TESLA and EMSS, for secure lossy multicast streams. TESLA (Timed Efficient Stream Loss-tolerant Authentication), offers sender authentication, strong loss robustness, high scalability and minimal overhead at the cost of loose initial time synchronization and slightly delayed authentication. EMSS (Efficient Multi-chained Stream Signature), provides nonrepudiation of origin, high loss resistance, and low overhead, at the cost of slightly delayed verification.

1,082 citations


Journal ArticleDOI
TL;DR: This work proposes an algorithm to optimally estimate the overall distortion of decoder frame reconstruction due to quantization, error propagation, and error concealment and recursively computes the total decoder distortion at pixel level precision to accurately account for spatial and temporal error propagation.
Abstract: Resilience to packet loss is a critical requirement in predictive video coding for transmission over packet-switched networks, since the prediction loop propagates errors and causes substantial degradation in video quality. This work proposes an algorithm to optimally estimate the overall distortion of decoder frame reconstruction due to quantization, error propagation, and error concealment. The method recursively computes the total decoder distortion at pixel level precision to accurately account for spatial and temporal error propagation. The accuracy of the estimate is demonstrated via simulation results. The estimate is integrated into a rate-distortion (RD)-based framework for optimal switching between intra-coding and inter-coding modes per macroblock. The cost in computational complexity is modest. The framework is further extended to optimally exploit feedback/acknowledgment information from the receiver/network. Simulation results both with and without a feedback channel demonstrate that precise distortion estimation enables the coder to achieve substantial and consistent gains in PSNR over known state-of-the-art RD- and non-RD-based mode switching methods.

717 citations


Journal ArticleDOI
28 Aug 2000
TL;DR: This paper presents a two-year study of Internet routing convergence through the experimental instrumentation of key portions of the Internet infrastructure, including both passive data collection and fault-injection machines at major Internet exchange points, and describes several unexpected properties of convergence.
Abstract: This paper examines the latency in Internet path failure, failover and repair due to the convergence properties of inter-domain routing. Unlike switches in the public telephony network which exhibit failover on the order of milliseconds, our experimental measurements show that inter-domain routers in the packet switched Internet may take tens of minutes to reach a consistent view of the network topology after a fault. These delays stem from temporary routing table oscillations formed during the operation of the BGP path selection process on Internet backbone routers. During these periods of delayed convergence, we show that end-to-end Internet paths will experience intermittent loss of connectivity, as well as increased packet loss and latency. We present a two-year study of Internet routing convergence through the experimental instrumentation of key portions of the Internet infrastructure, including both passive data collection and fault-injection machines at major Internet exchange points. Based on data from the injection and measurement of several hundred thousand inter-domain routing faults, we describe several unexpected properties of convergence and show that the measured upper bound on Internet inter-domain routing convergence delay is an order of magnitude slower than previously thought. Our analysis also shows that the upper theoretic computational bound on the number of router states and control messages exchanged during the process of BGP convergence is factorial with respect to the number of autonomous systems in the Internet. Finally, we demonstrate that much of the observed convergence delay stems from specific router vendor implementation decisions and ambiguity in the BGP specification.

542 citations


Proceedings ArticleDOI
26 Mar 2000
TL;DR: The extended model characterizes the expected value and distribution of TCP connection establishment and data transfer latency as a function of transfer size, round trip time, and packet loss rate, and it is shown that, unlike earlier steady-state models for TCP performance, the extended model describes connectionestablishment and dataTransfer latency under a range of packet loss conditions, including no loss.
Abstract: Several analytic models describe the steady-state throughput of bulk transfer TCP flows as a function of round trip time and packet loss rate. These models describe flows based on the assumption that they are long enough to sustain many packet losses. However, most TCP transfers across today's Internet are short enough to see few, if any, losses and consequently their performance is dominated by startup effects such as connection establishment and slow start. This paper extends the steady-state model proposed in Padhye et al. (1998), in order to capture these startup effects. The extended model characterizes the expected value and distribution of TCP connection establishment and data transfer latency as a function of transfer size, round trip time, and packet loss rate. Using simulations, controlled measurements of TCP transfers, and live Web measurements we show that, unlike earlier steady-state models for TCP performance, our extended model describes connection establishment and data transfer latency under a range of packet loss conditions, including no loss.

512 citations


Journal ArticleDOI
TL;DR: It is found that when optimizing for an exponential packet loss model with a mean loss rate of 20% and using a total rate of 0.2 bits per pixel on the Lenna image, good image quality can be obtained even when 40% of transmitted packets are lost.
Abstract: We present the unequal loss protection (ULP) framework in which unequal amounts of forward error correction are applied to progressive data to provide graceful degradation of image quality as packet losses increase. We develop a simple algorithm that can find a good assignment within the ULP framework. We use the set partitioning in hierarchical trees coder in this work, but our algorithm can protect any progressive compression scheme. In addition, we promote the use of a PMF of expected channel conditions so that our system can work with almost any model or estimate of packet losses. We find that when optimizing for an exponential packet loss model with a mean loss rate of 20% and using a total rate of 0.2 bits per pixel on the Lenna image, good image quality can be obtained even when 40% of transmitted packets are lost.

504 citations


Journal ArticleDOI
28 Aug 2000
TL;DR: A deterministic model of packet delay is described and used to derive both the packet pair property of FIFO-queueing networks and a new technique packet tailgating ) for actively measuring link bandwidths.
Abstract: We describe a deterministic model of packet delay and use it to derive both the packet pair [2] property of FIFO-queueing networks and a new technique packet tailgating) for actively measuring link bandwidths. Compared to previously known techniques, packet tailgating usually consumes less network bandwidth, does not rely on consistent behavior of routers handling ICMP packets, and does not rely on timely delivery of acknowledgments.

428 citations


Patent
29 Feb 2000
TL;DR: In this article, a system and method for managing a flow of packets through a network interface is presented, where a packet within a network flow is received, an operation code is generated for identifying whether the packet is suitable for a particular network interface function.
Abstract: A system and method are provided for managing a flow of packets through a network interface. A network flow is established for each datagram sent from a source entity to a destination entity. A flow key identifies the source and destination entities and is stored with information concerning validity of the flow, data sequencing within the flow datagram and how recently the flow was active. When a packet within a network flow is received, an operation code is generated for identifying whether the packet is suitable for a particular network interface function. One operation code may indicate that a packet contains data to be re-assembled with other flow data. Another operation code may indicate that a packet is unsuitable for data re-assembly. Another operation code may specify that the packet is a control packet, has no data, or was received out of order.

301 citations


Proceedings ArticleDOI
29 Dec 2000
TL;DR: In this article, the authors proposed a path diversity transmission system for video communication over lossy packet networks, where the system is composed of two subsystems: (1) multiple state video encoder/decoder and (2) a path-diversity transmission system.
Abstract: Video communication over lossy packet networks such as the Internet is hampered by limited bandwidth and packet loss. This paper presents a system for providing reliable video communication over these networks, where the system is composed of two subsystems: (1) multiple state video encoder/decoder and (2) a path diversity transmission system. Multiple state video coding combats the problem of error propagation at the decoder by coding the video into multiple independently decodable streams, each with its own prediction process and state. If one stream is lost the other streams can still be decoded to produce usable video, and furthermore, the correctly received streams provide bidirectional (previous and future) information that enables improved state recovery for the corrupted stream. This video coder is a form of multiple description coding (MDC), and its novelty lies in its use of information from the multiple streams to perform state recovery at the decoder. The path diversity transmission system explicitly sends different subsets of packets over different paths, as opposed to the default scenarios where the packets proceed along a single path, thereby enabling the end- to-end video application to effectively see an average path behavior. We refer to this as path diversity. Generally, seeing this average path behavior provides better performance than seeing the behavior of any individual random path. For example, the probability that all of the multiple paths are simultaneously congested is much less than the probability that a single path is congested. The resulting path diversity provides the multiple state video decoder with an appropriate virtual channel to assist in recovering from lost packets, and can also simplify system design, e.g. FEC design. We propose two architectures for achieving path diversity, and examine the effectiveness of path diversity in communicating video over a lossy packet network.

241 citations


Proceedings ArticleDOI
24 Sep 2000
TL;DR: This work describes a new carrier-sense multiple access (CSMA) protocol for multihop wireless networks, using multiple channels and a distributed channel selection scheme that provides a higher throughput and reduces the packet loss due to collisions.
Abstract: We describe a new carrier-sense multiple access (CSMA) protocol for multihop wireless networks, using multiple channels and a distributed channel selection scheme. The proposed protocol divides the available bandwidth into N channels where the transmitting station selects an appropriate channel for packet transmission. The selection criterion is based on the interference power measurements on the channels. We show via simulations that this multichannel CSMA protocol provides a higher throughput compared to its single channel counterpart by reducing the packet loss due to collisions.

222 citations


Patent
19 Apr 2000
TL;DR: In this paper, the authors describe a process of sending real-time information from a sender computer to a receiver computer coupled to the sender computer by a packet network wherein packets sometimes become lost.
Abstract: In one form of the invention, a process of sending real-time information from a sender computer to a receiver computer coupled to the sender computer by a packet network wherein packets sometimes become lost, includes steps of directing packets containing the real-time information from the sender computer by at least one path in the packet network to the receiver computer, and directing packets containing information dependent on the real-time information from the sender computer by at least one path deversity path in the packet network to the same receiver computer.

Patent
10 Aug 2000
TL;DR: In this article, a forward error correction (FEC) technique is proposed for interactive video transmission, which is based on the recovery from error spread using continuous updates (RESCU).
Abstract: Real-time interactive video transmission in the current Internet has mediocre quality because of high packet loss rates. Loss of packets belonging to a video frame is evident not only in the reduced quality of that frame but also in the propagation of that distortion to successive frames. This error propagation problem is inherent in any motion-based video codec because of the interdependence of encoded video frames. Since packet losses in the best-effort Internet environment cannot be prevented, minimizing the impact of these packet losses to the final video quality is important. A new forward error correction (FEC) technique effectively alleviates error propagation in the transmission of interactive video. The technique is based on a recently developed error recovery scheme called Recovery from Error Spread using Continuous Updates (RESCU). RESCU allows transport level recovery techniques previously known to be infeasible for interactive video transmission applications to be successfully used in such applications. The FEC technique can be very useful when the feedback channel from the receiver is highly limited, or transmission delay is high. Both simulation and Internet experiments indicate that the FEC technique effectively alleviates the error spread problem and is able to sustain much better video quality than H.261 or other conventional FEC schemes under various packet loss rates.

Patent
29 Feb 2000
TL;DR: In this article, a re-assembly engine reassembles data portions of multiple packets from a single communication flow, and header portions of re-assembled packets are stored in a header buffer.
Abstract: A network interface receives a packet from a network and transfers it to a host computer system. A header portion of the packet is parsed by a parser module to determine if the packet conforms to a predetermined protocol. A flow database is maintained by a flow database manager to reflect the creation, termination and activity of communication flows. A re-assembly engine re-assembles data portions of multiple packets from a single communication flow. Header portions of re-assembled packets are stored in a header buffer. When multiple packets in one flow are transferred to the host, a packet batching module enables their header portions to be processed collectively rather than being interspersed with other packets. A packet queue stores packets awaiting transfer to the host and a control queue stores information concerning the waiting packets. If the packet queue becomes saturated with packets, a random packet may be discarded.

Journal ArticleDOI
TL;DR: This paper describes a framework for admission control for a packet-based network where the decisions are taken by edge devices or end-systems, rather than resources within the network, and allows networks to be explicitly analyzed, and consequently engineered.
Abstract: This paper describes a framework for admission control for a packet-based network where the decisions are taken by edge devices or end-systems, rather than resources within the network. The decisions are based on the results of probe packets that the end-systems send through the network, and require only that resources apply a mark to packets in a way that is load dependent. One application example is the Internet, where marking information is fed back via an ECN bit, and we show how this approach allows a rich QoS framework for flows or streams. Our approach allows networks to be explicitly analyzed, and consequently engineered.

Journal ArticleDOI
TL;DR: The aim of the paper is to explain the basic p-cycle concept and its adaptation to both link and node restoration in the IP transport layer, and to outline certain initial results on the problem of optimized design of p- cycle based IP networks.
Abstract: We describe a novel restoration strategy called virtual protection cycles (p-cycles, patents pending) for extremely fast restoration in IP networks. Originally conceived for use in WDM and Sonet transport networks, we outline the adaption of the p-cycle concept to an IP environment. In an IP router-based network, p-cycles are implemented with virtual circuits techniques (such as an MPLS label switched path, or other means) to form closed logical loops that protect a number of IP links, or a node. In the event of failure, packets which would normally have been lost are encapsulated with a p-cycle IP address and reenter the routing table, which diverts them onto a protection cycle. They travel by normal forwarding or label switching along the p-cycle until they reach a node where the continuing route cost to the original destination is lower than that at the p-cycle entry node. Diverted packets are deencapsulated (dropped from the p-cycle) at that node and follow a normal (existing) route from there to their destination. Conventional routing protocols such as OSPF remain in place and operate as they do today, to develop a longer term global update to routing tables. Diversionary flows on the p-cycle inherently cease when the global routing update takes effect in response to the failed link or node. The p-cycle thus provides an immediate real-time detour, preventing packet loss, until conventional global routing reconvergence occurs. The aim of the paper is to explain the basic p-cycle concept and its adaptation to both link and node restoration in the IP transport layer, and to outline certain initial results on the problem of optimized design of p-cycle based IP networks.

Patent
András Veres1, Attila Farago1
07 Nov 2000
TL;DR: In this article, an end-to-end QoS metrics are provided for TCP connections based on the observation of packet flows at a single monitoring point, such as packet loss internally and externally to the monitoring point and detection of stalled periods and estimation of path delay.
Abstract: A method a system of identifying and determining degradation of the quality of service (QoS) perceived by a subscriber in a network such as the Internet. Traffic of individual applications of the subscriber and aggregate traffic of a subscriber are monitored, captured, and processed to produce QoS statistics. End-to-end QoS metrics are provided for TCP connections based on the observation of packet flows at a single monitoring point. The QoS metrics include, for example, packet loss internally and externally to the monitoring point, detection of stalled periods and estimation of path delay.

Journal ArticleDOI
TL;DR: A reliable multicast architecture that invokes active services at strategic locations inside the network to comprehensively address challenges such as feedback implosion, retransmission scoping, distributed loss recovery, and congestion control is presented.
Abstract: Scalability is of paramount importance in the design of reliable multicast transport protocols, and requires careful consideration of a number of problems such as feedback implosion, retransmission scoping, distributed loss recovery, and congestion control. In this article, we present a reliable multicast architecture that invokes active services at strategic locations inside the network to comprehensively address these challenges. Active services provide the ability to quickly and efficiently recover from loss at the point of loss. They also exploit the physical hierarchy for feedback aggregation and effective retransmission scoping with minimal router support. We present two protocols, one for packet loss recovery and another for congestion control, and describe an experimental testbed where these have been implemented. Analytical and experimental results are used to demonstrate that the active services architecture improves resource usage, reduces latency for loss recovery, and provides TCP-friendly congestion control.

Journal ArticleDOI
TL;DR: In this article, an analytical model is proposed in order to determine the number of converters needed to satisfy prefixed packet loss probability constraints in a bufferless packet optical switch employing the wavelength dimension for contention resolution.
Abstract: We propose an architecture for a bufferless packet optical switch employing the wavelength dimension for contention resolution. The optical packet switch is equipped with tunable wavelength converters shared among the input lines. An analytical model Is proposed in order to determine the number of converters needed to satisfy prefixed packet loss probability constraints. This analytical model very accurately fits with simulations results. A sensitivity analysis of the required number of converters as a function of the main system parameters (number of input and output lines, number of wavelengths, ...) and traffic parameters has been carried out. Making use of the introduced dimensioning procedure we have observed that the proposed architecture allows a saving in terms of employed number of converters with respect to the other architectures proposed in literature. Such a saving can reach about 95% of the number of converters.

Journal ArticleDOI
TL;DR: It is concluded that a clever design of the lower layers that preserve error correlations, naturally present on wireless links because of the fading behavior, could be an attractive alternative to the development or the use of more complex versions of TCP.
Abstract: The focus of this paper is to analyze the relative sensitivity of the bulk throughput performance of different versions of TCP, viz., OldTahoe, Tahoe, Reno, and New Reno, to channel errors that are correlated. We investigate the performance of a single wireless TCP connection in a local environment by modeling the correlated packet loss/error process (e.g., as induced by a multipath fading channel) as a first-order Markov chain. A major contribution of the paper is a unified analytical approach which allows the evaluation of the throughput performance of various versions of TCP. The main findings of this study are that 1) error correlations significantly affect the performance of TCP, and in particular may result in considerably better performance for Tahoe and NewReno; and 2) over slowly fading channels which are characterized by significant channel memory, Tahoe performs as well as NewReno. This leads us to conclude that a clever design of the lower layers that preserve error correlations, naturally present on wireless links because of the fading behavior, could be an attractive alternative to the development or the use of more complex versions of TCP.

Proceedings ArticleDOI
26 Mar 2000
TL;DR: The proposed methodology stems from a Markovian model of a single TCP source, and eventually considers the superposition and interaction of several such sources using standard queueing analysis techniques, and allows the evaluation of such performance indices as throughput, queueing delay and packet loss of TCP flows.
Abstract: In this paper, we outline a methodology that can be applied to model the behavior of TCP flows. The proposed methodology stems from a Markovian model of a single TCP source, and eventually considers the superposition and interaction of several such sources using standard queueing analysis techniques. Our approach allows the evaluation of such performance indices as throughput, queueing delay and packet loss of TCP flows. The results obtained through our model are validated by means of simulation, under several topology and traffic settings.

Patent
27 Sep 2000
TL;DR: In this paper, a system and method for transmitting data in a data communications network, using a transmission control protocol, to provide reduced acknowledgment control traffic, error recovery and congestion control is presented.
Abstract: A system and method for transmitting data in a data communications network, using a transmission control protocol, to provide reduced acknowledgment control traffic, error recovery and congestion control. A communications link is established between a transmitter and a receiver. Setting the communications link includes setting a network congestion window to an initial length. A sequence, or stream, of data packets is sent from the transmitter to the receiver. The receiver detects any missing packets, by examining the sequence numbers of the incoming packets, and sends negative acknowledgments, generally no more than four, to the transmitter identifying the missing data packet. When the transmitter receives a negative acknowledgment, it decreases the length of the congestion window, and re-transmits the missing packet. Detection and use of round-trip time, re-transmission time-out are provided.

Proceedings ArticleDOI
26 Mar 2000
TL;DR: It is shown that the source rates tend to be distributed in order to maximize an objective function called F/sub A//sup h/ ("F/ sub A//Sup h/ fairness"), which provides some insight into the distribution of rates, and hence of packet loss ratios, which can be expected in a given network with a number of competing TCP or TCP-friendly sources.
Abstract: Consider a network with an arbitrary topology and arbitrary communication delays, in which congestion control is based on additive-increase and multiplicative-decrease. We show that the source rates tend to be distributed in order to maximize an objective function called F/sub A//sup h/ ("F/sub A//sup h/ fairness"). We derive this result under the assumption of rate proportional negative feedback and for the regime of rare negative feedback. This applies to TCP in moderately loaded networks, and to those TCP implementations that are designed to interpret multiple packet losses within one RTT as a single congestion indication and do not rely on re-transmission timeout. This result provides some insight into the distribution of rates, and hence of packet loss ratios, which can be expected in a given network with a number of competing TCP or TCP-friendly sources. We validate our findings by analyzing a multiple-bottleneck scenario, and comparing with previous results (Floyd, 1991, Mathis et al, 1997) and an extensive numerical simulation with realistic parameter settings. We apply F/sub A//sup h/ fairness to gain a more accurate understanding of the bias of TCP against long round-trip times.

Journal ArticleDOI
TL;DR: This work considers the cases of a single and multiplexed traffic streams and derives the exact packet-loss rate (PLR) due to buffer overflow at the sender side of the wireless link and obtains a good approximation using the Chernoff-dominant eigenvalue (CDE) approach.
Abstract: Providing quality-of-service (QoS) guarantees over wireless packet networks poses a host of technical challenges that are not present in wireline networks. One of the key issues is how to account for the characteristics of the time-varying wireless channel and for the impact of link-layer error control in the provisioning of packet-level QoS. We accommodate both aspects in analyzing the packet-loss performance over a wireless link. We consider the cases of a single and multiplexed traffic streams. The link capacity fluctuates according to a fluid version of Gilbert-Elliott channel model. Traffic sources are modeled as on-off fluid processes. For the single-stream case, we derive the exact packet-loss rate (PLR) due to buffer overflow at the sender side of the wireless link. We also obtain a closed-form approximation for the corresponding wireless effective bandwidth. In the case of multiplexed streams, we obtain a good approximation for the PLR using the Chernoff-dominant eigenvalue (CDE) approach. Our analysis is then used to study the optimal forward error correction code rate that guarantees a given PLR while minimizing the allocated bandwidth. Numerical results and simulations are used to verify the adequacy of our analysis and to study the impact of error control on the allocation of bandwidth for guaranteed packet-loss performance.

Proceedings ArticleDOI
26 Mar 2000
TL;DR: A number of novel playout buffer algorithms are proposed which provide this coupling, and their effectiveness is demonstrated through simulations based on both network models and real network traces.
Abstract: Transport of real-time voice traffic on the Internet is difficult due to packet loss and jitter. Packet loss is handled primarily through a variety of different forward error correction (FEC) algorithms and local repair at the receiver. Jitter is compensated for by means of adaptive playout buffer algorithms at the receiver. Traditionally, these two mechanisms have been investigated in isolation. In this paper, we show the interactions between adaptive playout buffer algorithms and FEC, and demonstrate the need for coupling. We propose a number of novel playout buffer algorithms which provide this coupling, and demonstrate their effectiveness through simulations based on both network models and real network traces.

Patent
30 Nov 2000
TL;DR: In this article, the optimized character of FIFO for sequential transfer is maintained, while particular types of packets are processed out of order to achieve minimum latency and maximum data security in an intelligent network interface card.
Abstract: Support for priority and IP security packets, and other protocols at the network interface level and in conjunction with FIFO-based packet buffers is provided by allowing out of order processing of certain packets in the FIFO The optimized character of FIFO for sequential transfer is maintained, while particular types of packets are processed out of order to achieve minimum latency and maximum data security in an intelligent network interface card (10) A buffer (15) stores data packets in an order of receipt Logic is included in the network interface to transfer packets out of the buffer (15) according to the order of receipt, and according to the respective packet types so that packets having a particular packet type are transferred out of the order of receipt relative to packets having other packet types

Patent
29 Nov 2000
TL;DR: In this paper, a network system for actively controlling congestion to optimize throughput is provided, which includes a sending host which is configured to send packet traffic at a set rate and a receiving end which is the recipient of the packet traffic and also generates acknowledgment packets back to the sending host.
Abstract: A network system for actively controlling congestion to optimize throughput is provided. The network system includes a sending host which is configured to send packet traffic at a set rate. The network system also includes a sending switch for receiving the packet traffic. The sending switch includes an input buffer for receiving the packet traffic at the set rate where the input buffer is actively monitored to ascertain a capacity level. The sending switch also includes code for setting a probability factor that is correlated to the capacity level where the probability factor increases as the capacity level increases and decreases as the capacity level decreases. The sending switch also has code for randomly generating a value where the value is indicative of whether packets being sent by the sending switch are to be marked with a congestion indicator. The sending switch also includes transmit code that forwards the packet traffic out of the sending switch where the packet traffic includes one of marked packets and unmarked packets. The network system also has a receiving end which is the recipient of the packet traffic and also generates acknowledgment packets back to the sending host where the acknowledgment packets are marked with the congestion indicator when receiving marked packets and are not marked with the congestion indicator when receiving unmarked packets. In another example, the sending host is configured to monitor the acknowledgment packets and to adjust the set rate based on whether the acknowledgment packets are marked with the congestion indicator. In a further example, the set rate is decreased every time one of the marked packets is detected and increased when no marked packets are detected per round trip time (PRTT).

Patent
29 Sep 2000
TL;DR: In this paper, a method and apparatus for adaptively enforcing Quality of Service (QoS) policies for one or more flows of packets in a packet-switched network based on network feedback information is presented.
Abstract: A method and apparatus for adaptively enforcing Quality of Service (QoS) policies for one or more flows of packets in a packet-switched network based on network feedback information. In one aspect, packets of a first group of flows are assigned to a first service level. Then-current interface congestion information for network traffic that is mapped to the first service level and that is passing through an interface of a network device in the network is received. Based on the then-current interface congestion information one or more flows from the first group of flows are selected. Packets from the one or more flows are then assigned to a second service level.

Patent
28 Mar 2000
TL;DR: In this article, the authors propose an apparatus for and a method of dynamically prioritizing packets over a packet based network, where packets are dynamically prioritized on the basis of their "time to live" in the network as they travel from one network entity to another.
Abstract: An apparatus for and a method of dynamically prioritizing packets over a packet based network. Packets are dynamically prioritized on the basis of their ‘time to live’ in the network as they travel from one network entity to another. Packets are assigned a priority in accordance with how ‘old’ or ‘young’ they are. Packets with a relatively long time left to live are assigned lower priority then those with relatively little time left to live. A time to live (TTL) field is added to the packet as it travels from one network entity to another. The contents of the time to live (TTL) field represents how ‘young’ or ‘old’ the packet is and conveys the time left before the packet is no longer of any use. Each network entity that receives the packet with a TTL field, subtracts from it the time the packet spends passing through that entity. The field decreases as it hops from network entity to entity until it reaches its destination or is discarded.

Patent
Heikki Salovuori1
29 Nov 2000
TL;DR: In this article, the authors propose a method of routing calls in a telecommunication system comprising a mobile communication system comprising at least one mobile services switching center and at least three base stations connected thereto, and mobile stations where from a tele-communication connection is established to the mobile service switching centre through the base station by using a first predetermined signalling protocol on a radio connection.
Abstract: A method of routing calls in a telecommunication system comprising a mobile communication system comprising at least one mobile services switching centre and at least one base station connected thereto, and mobile stations wherefrom a telecommunication connection is established to the mobile services switching centre through the base station by using a first predetermined signalling protocol on a radio connection. The telecommunication system further comprises a packet data network wherein data to be transmitted is transmitted in data packets, according to a protocol specified for the packet data network, a server communicating with the mobile services switching centre and the packet data network, wherefrom a telecommunication connection is established through the packet data network and the server to the mobile switching centre. In the server, messages according to the first signalling protocol supplied from the mobile services switching centre are as such arranged into data packets according to the packet data network protocol to be forwarded through the packet data network to the terminal. Furthermore in the server, messages according to the first signalling protocol arranged as such in the data packets according to the packet data network protocol and supplied from the packet data network are decompressed to be forwarded to the mobile switching centre. The terminal transmits and receives the messages according to the first signalling protocol arranged as such into the data packets according to the packet data network protocol and emulates a mobile station using the first signalling protocol.

Proceedings ArticleDOI
28 Mar 2000
TL;DR: In this paper, the authors consider the problem of joint source/channel coding of real-time sources, such as audio and video, for the purpose of multicasting over the Internet.
Abstract: We consider the problem of joint source/channel coding of real-time sources, such as audio and video, for the purpose of multicasting over the Internet. The sender injects into the network multiple source layers and multiple channel (parity) layers, some of which are delayed relative to the source. Each receiver subscribes to the number of source layers and the number of channel layers that optimizes the source-channel rate allocation for that receiver's available bandwidth and packet loss probability. We augment this layered FEC system with layered ARQ. Although feedback is normally problematic in broadcast situations, ARQ is simulated by having the receivers subscribe and unsubscribe to the delayed channel coding layers to receive missing information. This pseudo-ARQ scheme avoids an implosion of repeat requests at the sender, and is scalable to an unlimited number of receivers. We show gains of up to 18 dB on channels with 20% loss over systems without error control, and additional gains of up to 13 dB when FEC is augmented by pseudo-ARQ in a hybrid system. The hybrid system is controlled by an optimal policy for a Markov decision process.