diff --git a/test/db.json b/test/db.json index ec83c2afe..b49d140c4 100644 --- a/test/db.json +++ b/test/db.json @@ -54,7 +54,7 @@ { "_id_": 3, "title": "A Simulation Study of IP Switching", - "abstract": "Recently there has been much interest in combining the speed of layer-2 switching with the features of layer-3 routing. This has been prompted by numerous proposals, including: IP Switching [1], Tag Switching [2], ARIS [3], CSR [4], and IP over ATM [5]. In this application, we study IP Switching and evaluate the performance claims made by Newman et al in [1] and [6]. In particular, using ten network traces, we study how well IP Switching performs with traffic found in campus, corporate, and Internet Service Provider (ISP) environments. Our main finding is that IP Switching will lead to a high proportion of datagrams that are switched; over 75% in all of the environments we studied. We also investigate the effects that different flow classifiers and various timer values have on performance, and note that some choices can result in a large VC space requirement. Finally, we present recommendations for the flow classifier and timer values, as a function of the VC space of the switch and the network environment being served.", + "abstract": "Recently there has been much interest in combining the speed of layer-2 switching with the features of layer-3 routing. This has been prompted by numerous proposals, including: IP Switching [1], Tag Switching [2], ARIS [3], CSR [4], and IP over ATM [5]. In this paper, we study IP Switching and evaluate the performance claims made by Newman et al in [1] and [6]. In particular, using ten network traces, we study how well IP Switching performs with traffic found in campus, corporate, and Internet Service Provider (ISP) environments. Our main finding is that IP Switching will lead to a high proportion of datagrams that are switched; over 75% in all of the environments we studied. We also investigate the effects that different flow classifiers and various timer values have on performance, and note that some choices can result in a large VC space requirement. Finally, we present recommendations for the flow classifier and timer values, as a function of the VC space of the switch and the network environment being served.", "authors": [ {"name":"Steven Lin", "email": "sclin@leland.stanford.edu", "affiliation": "Stanford University", "contact": true}, {"name":"Nick McKeown", "email": "nickm@ee.stanford.edu", "affiliation": "Stanford University"} @@ -67,7 +67,7 @@ { "_id_": 4, "title": "Scalable High Speed IP Routing Lookups", - "abstract": "Internet address lookup is a challenging problem because of increasing routing table sizes, increased traffic, higher speed links, and the migration to 128 bit IPv6 addresses. IP routing lookup requires computing the best matching prefix, for which standard solutions like hashing were believed to be inapplicable. The best existing solution we know of, BSD radix tries, scales badly as IP moves to 128 bit addresses. Our application describes a new algorithm for best matching prefix using binary search on hash tables organized by prefix lengths. Our scheme scales very well as address and routing table sizes increase: independent of the table size, it requires a worst case time of log2(address bits) hash lookups. Thus only 5 hash lookups are needed for IPv4 and 7 for IPv6. We also introduce Mutating Binary Search and other optimizations that, for a typical IPv4 backbone router with over 33,000 entries, considerably reduce the average number of hashes to less than 2, of which one hash can be simplified to an indexed array access. We expect similar average case behavior for IPv6.", + "abstract": "Internet address lookup is a challenging problem because of increasing routing table sizes, increased traffic, higher speed links, and the migration to 128 bit IPv6 addresses. IP routing lookup requires computing the best matching prefix, for which standard solutions like hashing were believed to be inapplicable. The best existing solution we know of, BSD radix tries, scales badly as IP moves to 128 bit addresses. Our paper describes a new algorithm for best matching prefix using binary search on hash tables organized by prefix lengths. Our scheme scales very well as address and routing table sizes increase: independent of the table size, it requires a worst case time of log2(address bits) hash lookups. Thus only 5 hash lookups are needed for IPv4 and 7 for IPv6. We also introduce Mutating Binary Search and other optimizations that, for a typical IPv4 backbone router with over 33,000 entries, considerably reduce the average number of hashes to less than 2, of which one hash can be simplified to an indexed array access. We expect similar average case behavior for IPv6.", "authors": [ {"name":"Marcel Waldvogel", "email": "waldvogel@tik.ee.ethz.ch", "affiliation": "Computer Engineering and Networks Laboratory, ETH Zürich", "contact": true}, {"name":"George Varghese", "email": "varghese@ccrc.wustl.edu", "affiliation": "Computer and Communications Research Center, Washington University in St. Louis", "contact": true}, @@ -93,7 +93,7 @@ { "_id_": 6, "title": "Trace-Based Mobile Network Emulation", - "abstract": "Subjecting a mobile computing system to wireless network conditions that are realistic yet reproducible is a challenging problem. In this application, we describe a technique called trace modulation that re-creates the observed end-to-end characteristics of a real wireless network in a controlled and repeatable manner. Trace modulation is transparent to applications and accounts for all network traffic sent or received by the system under test. We present results that show that it is indeed capable of reproducing wireless network performance faithfully.", + "abstract": "Subjecting a mobile computing system to wireless network conditions that are realistic yet reproducible is a challenging problem. In this paper, we describe a technique called trace modulation that re-creates the observed end-to-end characteristics of a real wireless network in a controlled and repeatable manner. Trace modulation is transparent to applications and accounts for all network traffic sent or received by the system under test. We present results that show that it is indeed capable of reproducing wireless network performance faithfully.", "authors": [ {"name":"Brian Noble", "email": "bnoble@cs.cmu.edu", "affiliation": "Carnegie Mellon University, School of Computer Science", "contact": true}, {"name":"M. Satyanarayanan", "email": "satya@cs.cmu.edu", "affiliation": "Carnegie Mellon University, School of Computer Science"}, @@ -106,7 +106,7 @@ { "_id_": 7, "title": "Fair Scheduling in Wireless Packet Networks", - "abstract": "Fair scheduling of delay and rate-sensitive packet flows over a wireless channel is not addressed effectively by most contemporary wireline fair scheduling algorithms because of two unique characteristics of wireless media: (a) bursty channel errors, and (b) location-dependent channel capacity and errors. Besides, in packet cellular networks, the base station typically performs the task of packet scheduling for both downlink and uplink flows in a cell; however a base station has only a limited knowledge of the arrival processes of uplink flows.\n\nIn this application, we propose a new model for wireless fair scheduling based on an adaptation of fluid fair queueing to handle location-dependent error bursts. We describe an ideal wireless fair scheduling algorithm which provides a packetized implementation of the fluid model while assuming full knowledge of the current channel conditions. For this algorithm, we derive the worst-case throughput and delay bounds. Finally, we describe a practical wireless scheduling algorithm which approximates the ideal algorithm. Through simulations, we show that the algorithm achieves the desirable properties identified in the wireless fluid fair queueing model.", + "abstract": "Fair scheduling of delay and rate-sensitive packet flows over a wireless channel is not addressed effectively by most contemporary wireline fair scheduling algorithms because of two unique characteristics of wireless media: (a) bursty channel errors, and (b) location-dependent channel capacity and errors. Besides, in packet cellular networks, the base station typically performs the task of packet scheduling for both downlink and uplink flows in a cell; however a base station has only a limited knowledge of the arrival processes of uplink flows.\n\nIn this paper, we propose a new model for wireless fair scheduling based on an adaptation of fluid fair queueing to handle location-dependent error bursts. We describe an ideal wireless fair scheduling algorithm which provides a packetized implementation of the fluid model while assuming full knowledge of the current channel conditions. For this algorithm, we derive the worst-case throughput and delay bounds. Finally, we describe a practical wireless scheduling algorithm which approximates the ideal algorithm. Through simulations, we show that the algorithm achieves the desirable properties identified in the wireless fluid fair queueing model.", "authors": [ {"name":"Songwu Lu", "email": "slu@crhc.uiuc.edu", "affiliation": "UIUC", "contact": true}, {"name":"Vaduvur Bharghavan", "email": "bharghav@crhc.uiuc.edu", "affiliation": "UIUC"}, @@ -118,7 +118,7 @@ { "_id_": 8, "title": "Fast Restoration of Real-Time Communication Service from Component Failures in Multi-Hop Networks", - "abstract": "For many applications it is important to provide communication services with guaranteed timeliness and fault-tolerance at an acceptable level of overhead. In this application, we present a scheme for restoring real-time channels, each with guaranteed timeliness, from component failures in multi-hop networks. To ensure fast/guaranteed recovery, backup channels are set up a priori in addition to each primary channel. That is, a dependable real-time connection consists of a primary channel and one or more backup channels. If a primary channel fails, one of its backup channels is activated to become a new primary channel. We describe a protocol which provides an integrated solution to the failure-recovery problem (i.e., channel switching, resource re-allocation, ...). We also present a resource sharing method that significantly reduces the overhead of backup channels. The simulation results show that good coverage (in recovering from failures) can be achieved with about 30% degradation in network utilization under a reasonable failure condition. Moreover, the fault-tolerance level of each dependable connection can be controlled, independently of other connections, to reflect its criticality.", + "abstract": "For many applications it is important to provide communication services with guaranteed timeliness and fault-tolerance at an acceptable level of overhead. In this paper, we present a scheme for restoring real-time channels, each with guaranteed timeliness, from component failures in multi-hop networks. To ensure fast/guaranteed recovery, backup channels are set up a priori in addition to each primary channel. That is, a dependable real-time connection consists of a primary channel and one or more backup channels. If a primary channel fails, one of its backup channels is activated to become a new primary channel. We describe a protocol which provides an integrated solution to the failure-recovery problem (i.e., channel switching, resource re-allocation, ...). We also present a resource sharing method that significantly reduces the overhead of backup channels. The simulation results show that good coverage (in recovering from failures) can be achieved with about 30% degradation in network utilization under a reasonable failure condition. Moreover, the fault-tolerance level of each dependable connection can be controlled, independently of other connections, to reflect its criticality.", "authors": [ {"name":"Seungjae Han", "email": "sjhan@eecs.umich.edu", "affiliation": "University of Michigan", "contact": true}, {"name":"Kang G. Shin", "email": "kgshin@eecs.umich.edu", "affiliation": "University of Michigan"} @@ -140,7 +140,7 @@ { "_id_": 10, "title": "Active Bridging", - "abstract": "Active networks accelerate network evolution by permitting the network infrastructure to be programmable, on a per-user, per-packet, or other basis. This programmability must be balanced against the safety and security needs inherent in shared resources.This application describes the design, implementation, and performance of a new type of network element, an Active Bridge. The active bridge can be reprogrammed \"on the fly\", with loadable modules called switchlets. To demonstrate the use of the active property, we incrementally extend what is initially a programmable buffered repeater with switchlets into a self-learning bridge, and then a bridge supporting spanning tree algorithms. To demonstrate the agility that active networking gives, we show how it is possible to upgrade a network from an \"old\" protocol to a \"new\" protocol on-the-fly. Moreover, we are able to take advantage of information unavailable to the implementors of either protocol to validate the new protocol and fall back to the old protocol if an error is detected. This shows that the Active Bridge can protect itself from some algorithmic failures in loadable modules.Our approach to safety and security favors static checking and prevention over dynamic checks when possible. We rely on strong type checking in the Caml language for the loadable module infrastructure, and achieve respectable performance. The prototype implementation on a Pentium-based HP Netserver LS running Linux with 100 Mbps Ethernet LANS achieves ttcp throughput of 16 Mbps between two PCs running Linux, compared with 76 Mbps unbridged. Measured frame rates are in the neighborhood of 1800 frames per second.", + "abstract": "Active networks accelerate network evolution by permitting the network infrastructure to be programmable, on a per-user, per-packet, or other basis. This programmability must be balanced against the safety and security needs inherent in shared resources.This paper describes the design, implementation, and performance of a new type of network element, an Active Bridge. The active bridge can be reprogrammed \"on the fly\", with loadable modules called switchlets. To demonstrate the use of the active property, we incrementally extend what is initially a programmable buffered repeater with switchlets into a self-learning bridge, and then a bridge supporting spanning tree algorithms. To demonstrate the agility that active networking gives, we show how it is possible to upgrade a network from an \"old\" protocol to a \"new\" protocol on-the-fly. Moreover, we are able to take advantage of information unavailable to the implementors of either protocol to validate the new protocol and fall back to the old protocol if an error is detected. This shows that the Active Bridge can protect itself from some algorithmic failures in loadable modules.Our approach to safety and security favors static checking and prevention over dynamic checks when possible. We rely on strong type checking in the Caml language for the loadable module infrastructure, and achieve respectable performance. The prototype implementation on a Pentium-based HP Netserver LS running Linux with 100 Mbps Ethernet LANS achieves ttcp throughput of 16 Mbps between two PCs running Linux, compared with 76 Mbps unbridged. Measured frame rates are in the neighborhood of 1800 frames per second.", "authors": [ {"name":"D. Scott Alexander", "email": "salex@dsl.cis.upenn.edu", "affiliation": "University of Pennsylvania", "contact": true}, {"name":"Marianne Shaw", "email": "marianne@dsl.cis.upenn.edu", "affiliation": "University of Pennsylvania"}, @@ -153,7 +153,7 @@ { "_id_": 11, "title": "Internet Routing Instability", - "abstract": "This application examines the network inter-domain routing information exchanged between backbone service providers at the major U.S. public Internet exchange points. Internet routing instability, or the rapid fluctuation of network reachability information, is an important problem currently facing the Internet engineering community. High levels of network instability can lead to packet loss, increased network latency and time to convergence. At the extreme, high levels of routing instability have lead to the loss of internal connectivity in wide-area, national networks. In this application, we describe several unexpected trends in routing instability, and examine a number of anomalies and pathologies observed in the exchange of inter-domain routing information. The analysis in this application is based on data collected from BGP routing messages generated by border routers at five of the Internet core's public exchange points during a nine month period. We show that the volume of these routing updates is several orders of magnitude more than expected and that the majority of this routing information is redundant, or pathological. Furthermore, our analysis reveals several unexpected trends and ill-behaved systematic properties in Internet routing. We finally posit a number of explanations for these anomalies and evaluate their potential impact on the Internet infrastructure.", + "abstract": "This paper examines the network inter-domain routing information exchanged between backbone service providers at the major U.S. public Internet exchange points. Internet routing instability, or the rapid fluctuation of network reachability information, is an important problem currently facing the Internet engineering community. High levels of network instability can lead to packet loss, increased network latency and time to convergence. At the extreme, high levels of routing instability have lead to the loss of internal connectivity in wide-area, national networks. In this paper, we describe several unexpected trends in routing instability, and examine a number of anomalies and pathologies observed in the exchange of inter-domain routing information. The analysis in this paper is based on data collected from BGP routing messages generated by border routers at five of the Internet core's public exchange points during a nine month period. We show that the volume of these routing updates is several orders of magnitude more than expected and that the majority of this routing information is redundant, or pathological. Furthermore, our analysis reveals several unexpected trends and ill-behaved systematic properties in Internet routing. We finally posit a number of explanations for these anomalies and evaluate their potential impact on the Internet infrastructure.", "authors": [ {"name":"Craig Labovitz", "email": "labovit@eecs.umich.edu", "affiliation": "University of Michigan Department of Electrical Engineering and Computer Science", "contact": true}, {"name":"G. Robert Malan", "email": "rmalan@eecs.umich.edu", "affiliation": "University of Michigan Department of Electrical Engineering and Computer Science", "contact": true}, @@ -165,7 +165,7 @@ { "_id_": 12, "title": "Dynamics of Random Early Detection", - "abstract": "In this application we evaluate the effectiveness of Random Early Detection (RED) over traffic types categorized as non-adaptive, fragile and robust, according to their responses to congestion. We point out that RED allows unfair bandwidth sharing when a mixture of the three traffic types shares a link. This unfairness is caused by the fact that at any given time RED imposes the same loss rate on all flows, regardless of their bandwidths.\n\nWe propose Fair Random Early Drop (FRED), a modified version of RED. FRED uses per-active-flow accounting to impose on each flow a loss rate that depends on the flow's buffer use.\n\nWe show that FRED provides better protection than RED for adaptive (fragile and robust) flows. In addition, FRED is able to isolate non-adaptive greedy traffic more effectively. Finally, we present a \"two-packet-buffer\" gateway mechanism to support a large number of flows without incurring additional queueing delays inside the network. These improvements are demonstrated by simulations of TCP and UDP traffic.\n\nFRED does not make any assumptions about queueing architecture; it will work with a FIFO gateway. FRED's per-active-flow accounting uses memory in proportion to the total number of buffers used: a FRED gateway maintains state only for flows for which it has packets buffered, not for all flows that traverse the gateway.", + "abstract": "In this paper we evaluate the effectiveness of Random Early Detection (RED) over traffic types categorized as non-adaptive, fragile and robust, according to their responses to congestion. We point out that RED allows unfair bandwidth sharing when a mixture of the three traffic types shares a link. This unfairness is caused by the fact that at any given time RED imposes the same loss rate on all flows, regardless of their bandwidths.\n\nWe propose Fair Random Early Drop (FRED), a modified version of RED. FRED uses per-active-flow accounting to impose on each flow a loss rate that depends on the flow's buffer use.\n\nWe show that FRED provides better protection than RED for adaptive (fragile and robust) flows. In addition, FRED is able to isolate non-adaptive greedy traffic more effectively. Finally, we present a \"two-packet-buffer\" gateway mechanism to support a large number of flows without incurring additional queueing delays inside the network. These improvements are demonstrated by simulations of TCP and UDP traffic.\n\nFRED does not make any assumptions about queueing architecture; it will work with a FIFO gateway. FRED's per-active-flow accounting uses memory in proportion to the total number of buffers used: a FRED gateway maintains state only for flows for which it has packets buffered, not for all flows that traverse the gateway.", "authors": [ {"name":"Dong Lin", "affiliation": "Harvard University"}, {"name":"Robert Morris", "email": "rtm@eecs.harvard.edu", "affiliation": "Harvard University", "contact": true} @@ -187,7 +187,7 @@ { "_id_": 14, "title": "Network Performance Effects of HTTP/1.1, CSS1, and PNG", - "abstract": "We describe our investigation of the effect of persistent connections, pipelining and link level document compression on our client and server HTTP implementations. A simple test setup is used to verify HTTP/1.1's design and understand HTTP/1.1 implementation strategies. We present TCP and real time performance data between the libwww robot [27] and both the W3C's Jigsaw [28] and Apache [29] HTTP servers using HTTP/1.0, HTTP/1.1 with persistent connections, HTTP/1.1 with pipelined requests, and HTTP/1.1 with pipelined requests and deflate data compression [22]. We also investigate whether the TCP Nagle algorithm has an effect on HTTP/1.1 performance. While somewhat artificial and possibly overstating the benefits of HTTP/1.1, we believe the tests and results approximate some common behavior seen in browsers. The results confirm that HTTP/1.1 is meeting its major design goals. Our experience has been that implementation details are very important to achieve all of the benefits of HTTP/1.1.\n\nFor all our tests, a pipelined HTTP/1.1 implementation outperformed HTTP/1.0, even when the HTTP/1.0 implementation used multiple connections in parallel, under all network environments tested. The savings were at least a factor of two, and sometimes as much as a factor of ten, in terms of packets transmitted. Elapsed time improvement is less dramatic, and strongly depends on your network connection.Some data is presented showing further savings possible by changes in Web content, specifically by the use of CSS style sheets [10], and the more compact PNG [20] image representation, both recent recommendations of W3C. Time did not allow full end to end data collection on these cases. The results show that HTTP/1.1 and changes in Web content will have dramatic results in Internet and Web performance as HTTP/1.1 and related technologies deploy over the near future. Universal use of style sheets, even without deployment of HTTP/1.1, would cause a very significant reduction in network traffic.This application does not investigate further performance and network savings enabled by the improved caching facilities provided by the HTTP/1.1 protocol, or by sophisticated use of range requests.", + "abstract": "We describe our investigation of the effect of persistent connections, pipelining and link level document compression on our client and server HTTP implementations. A simple test setup is used to verify HTTP/1.1's design and understand HTTP/1.1 implementation strategies. We present TCP and real time performance data between the libwww robot [27] and both the W3C's Jigsaw [28] and Apache [29] HTTP servers using HTTP/1.0, HTTP/1.1 with persistent connections, HTTP/1.1 with pipelined requests, and HTTP/1.1 with pipelined requests and deflate data compression [22]. We also investigate whether the TCP Nagle algorithm has an effect on HTTP/1.1 performance. While somewhat artificial and possibly overstating the benefits of HTTP/1.1, we believe the tests and results approximate some common behavior seen in browsers. The results confirm that HTTP/1.1 is meeting its major design goals. Our experience has been that implementation details are very important to achieve all of the benefits of HTTP/1.1.\n\nFor all our tests, a pipelined HTTP/1.1 implementation outperformed HTTP/1.0, even when the HTTP/1.0 implementation used multiple connections in parallel, under all network environments tested. The savings were at least a factor of two, and sometimes as much as a factor of ten, in terms of packets transmitted. Elapsed time improvement is less dramatic, and strongly depends on your network connection.Some data is presented showing further savings possible by changes in Web content, specifically by the use of CSS style sheets [10], and the more compact PNG [20] image representation, both recent recommendations of W3C. Time did not allow full end to end data collection on these cases. The results show that HTTP/1.1 and changes in Web content will have dramatic results in Internet and Web performance as HTTP/1.1 and related technologies deploy over the near future. Universal use of style sheets, even without deployment of HTTP/1.1, would cause a very significant reduction in network traffic.This paper does not investigate further performance and network savings enabled by the improved caching facilities provided by the HTTP/1.1 protocol, or by sophisticated use of range requests.", "authors": [ {"name":"Henrik Frystyk Nielsen", "email": "frystyk@w3.org", "affiliation": "World Wide Web Consortium", "contact": true}, {"name":"James Gettys", "email": "jg@pa.dec.com", "affiliation": "Visiting Scientist, World Wide Web Consortium; Digital Equipment Corporation", "contact": true}, @@ -212,7 +212,7 @@ { "_id_": 16, "title": "Potential Benefits of Delta Encoding and Data Compression for HTTP", - "abstract": "Caching in the World Wide Web currently follows a naive model, which assumes that resources are referenced many times between changes. The model also provides no way to update a cache entry if a resource does change, except by transferring the resource's entire new value. Several previous applications have proposed updating cache entries by transferring only the differences, or \"delta,\" between the cached entry and the current value.\n\nIn this application, we make use of dynamic traces of the full contents of HTTP messages to quantify the potential benefits of delta-encoded responses. We show that delta encoding can provide remarkable improvements in response size and response delay for an important subset of HTTP content types. We also show the added benefit of data compression, and that the combination of delta encoding and data compression yields the best results.\n\nWe propose specific extensions to the HTTP protocol for delta encoding and data compression. These extensions are compatible with existing implementations and specifications, yet allow efficient use of a variety of encoding techniques.", + "abstract": "Caching in the World Wide Web currently follows a naive model, which assumes that resources are referenced many times between changes. The model also provides no way to update a cache entry if a resource does change, except by transferring the resource's entire new value. Several previous papers have proposed updating cache entries by transferring only the differences, or \"delta,\" between the cached entry and the current value.\n\nIn this paper, we make use of dynamic traces of the full contents of HTTP messages to quantify the potential benefits of delta-encoded responses. We show that delta encoding can provide remarkable improvements in response size and response delay for an important subset of HTTP content types. We also show the added benefit of data compression, and that the combination of delta encoding and data compression yields the best results.\n\nWe propose specific extensions to the HTTP protocol for delta encoding and data compression. These extensions are compatible with existing implementations and specifications, yet allow efficient use of a variety of encoding techniques.", "authors": [ {"name":"Jeffrey C. Mogul", "email": "mogul@wrl.dec.com", "affiliation": "Digital Equipment Corporation Western Research Laboratory", "contact": true}, {"name":"Fred Douglis", "email": "douglis@research.att.com", "affiliation": "AT&T Labs - Research"}, @@ -225,7 +225,7 @@ { "_id_": 17, "title": "Network Text Editor (NTE): A Scalable Shared Text Editor for the MBone", - "abstract": "IP Multicast, Lightweight Sessions and Application Level Framing provide guidelines by which multimedia conferencing tools can be designed, but they do not provide specific solutions. In this application, we use these design principles to guide the design of a multicast based shared editor, and examine the consequences of taking a loose consistency approach to achieve good performance in the face of network failures and losses.", + "abstract": "IP Multicast, Lightweight Sessions and Application Level Framing provide guidelines by which multimedia conferencing tools can be designed, but they do not provide specific solutions. In this paper, we use these design principles to guide the design of a multicast based shared editor, and examine the consequences of taking a loose consistency approach to achieve good performance in the face of network failures and losses.", "authors": [ {"name":"Mark Handley", "email": "mjh@isi.edu", "affiliation": "USC Information Sciences Institute", "contact": true}, {"name":"Jon Crowcroft", "email": "jon@cs.ucl.ac.uk", "affiliation": "University College London", "contact": true} @@ -236,7 +236,7 @@ { "_id_": 18, "title": "Consistent Overhead Byte Stuffing", - "abstract": "Byte stuffing is a process that transforms a sequence of data bytes that may contain 'illegal' or 'reserved' values into a potentially longer sequence that contains no occurrences of those values. The extra length is referred to in this application as the overhead of the algorithm.\n\nTo date, byte stuffing algorithms, such as those used by SLIP [RFC1055], PPP [RFC1662] and AX.25 [ARRL84], have been designed to incur low average overhead but have made little effort to minimize worst case overhead.\n\nSome increasingly popular network devices, however, care more about the worst case. For example, the transmission time for ISM-band packet radio transmitters is strictly limited by FCC regulation. To adhere to this regulation, the practice is to set the maximum packet size artificially low so that no packet, even after worst case overhead, can exceed the transmission time limit.\n\nThis application presents a new byte stuffing algorithm, called Consistent Overhead Byte Stuffing (COBS), that tightly bounds the worst case overhead. It guarantees in the worst case to add no more than one byte in 254 to any packet. Furthermore, the algorithm is computationally cheap, and its average overhead is very competitive with that of existing algorithms.", + "abstract": "Byte stuffing is a process that transforms a sequence of data bytes that may contain 'illegal' or 'reserved' values into a potentially longer sequence that contains no occurrences of those values. The extra length is referred to in this paper as the overhead of the algorithm.\n\nTo date, byte stuffing algorithms, such as those used by SLIP [RFC1055], PPP [RFC1662] and AX.25 [ARRL84], have been designed to incur low average overhead but have made little effort to minimize worst case overhead.\n\nSome increasingly popular network devices, however, care more about the worst case. For example, the transmission time for ISM-band packet radio transmitters is strictly limited by FCC regulation. To adhere to this regulation, the practice is to set the maximum packet size artificially low so that no packet, even after worst case overhead, can exceed the transmission time limit.\n\nThis paper presents a new byte stuffing algorithm, called Consistent Overhead Byte Stuffing (COBS), that tightly bounds the worst case overhead. It guarantees in the worst case to add no more than one byte in 254 to any packet. Furthermore, the algorithm is computationally cheap, and its average overhead is very competitive with that of existing algorithms.", "authors": [ {"name":"Stuart Cheshire", "email": "cheshire@cs.stanford.edu", "affiliation": "Computer Science Department, Stanford University", "contact": true}, {"name":"Mary Baker", "email": "mgbaker@cs.stanford.edu", "affiliation": "Computer Science Department, Stanford University"} @@ -247,7 +247,7 @@ { "_id_": 19, "title": "A Flow-Based Approach to Datagram Security", - "abstract": "Datagram services provide a simple, flexible, robust, and scalable communication abstraction; their usefulness has been well demonstrated by the success of IP, UDP, and RPC. Yet, the overwhelming majority of network security protocols that have been proposed are geared towards connection-oriented communications. The few that do cater to datagram communications tend to either rely on long term host-pair keying or impose a session-oriented (i.e., requiring connection setup) semantics.\n\nSeparately, the concept of flows has received a great deal of attention recently, especially in the context of routing and QoS. A flow characterizes a sequence of datagrams sharing some pre-defined attributes. In this application, we advocate the use of flows as a basis for structuring secure datagram communications. We support this by proposing a novel protocol for datagram security based on flows. Our protocol achieves zero-message keying, thus preserving the connectionless nature of datagram, and makes use of soft state, thus providing the efficiency of session-oriented schemes. We have implemented an instantiation for IP in the 4.4BSD kernel, and we provide a description of our implementation along with performance results.", + "abstract": "Datagram services provide a simple, flexible, robust, and scalable communication abstraction; their usefulness has been well demonstrated by the success of IP, UDP, and RPC. Yet, the overwhelming majority of network security protocols that have been proposed are geared towards connection-oriented communications. The few that do cater to datagram communications tend to either rely on long term host-pair keying or impose a session-oriented (i.e., requiring connection setup) semantics.\n\nSeparately, the concept of flows has received a great deal of attention recently, especially in the context of routing and QoS. A flow characterizes a sequence of datagrams sharing some pre-defined attributes. In this paper, we advocate the use of flows as a basis for structuring secure datagram communications. We support this by proposing a novel protocol for datagram security based on flows. Our protocol achieves zero-message keying, thus preserving the connectionless nature of datagram, and makes use of soft state, thus providing the efficiency of session-oriented schemes. We have implemented an instantiation for IP in the 4.4BSD kernel, and we provide a description of our implementation along with performance results.", "authors": [ {"name": "Suvo Mittra", "email": "suvo@cs.stanford.edu", "affiliation": "Stanford University", "contact": true}, {"name": "Thomas Y.C. Woo", "email": "woo@research.bell-labs.com", "affiliation": "Bell Laboratories"} @@ -258,7 +258,7 @@ { "_id_": 20, "title": "Best-Effort versus Reservations: A Simple Comparative Analysis", - "abstract": "Using a simple analytical model, this application addresses the following question: Should the Internet retain its best-effort-only architecture, or should it adopt one that is reservation-capable? We characterize the differences between reservation-capable and best-effort-only networks in terms of application performance and total welfare. Our analysis does not yield a definitive answer to the question we pose, since it would necessarily depend on unknowable factors such as the future cost of network bandwidth and the nature of the future traffic load. However, our model does reveal some interesting phenomena. First, in some circumstances, the amount of incremental bandwidth needed to make a best-effort-only network perform as well as a reservation capable one diverges as capacity increases. Second, in some circumstances reservation-capable networks retain significant advantages over best-effort-only networks, no matter how cheap bandwidth becomes. Lastly, we find bounds on the maximum performance advantage a reservation-capable network can achieve over best effort architectures.", + "abstract": "Using a simple analytical model, this paper addresses the following question: Should the Internet retain its best-effort-only architecture, or should it adopt one that is reservation-capable? We characterize the differences between reservation-capable and best-effort-only networks in terms of application performance and total welfare. Our analysis does not yield a definitive answer to the question we pose, since it would necessarily depend on unknowable factors such as the future cost of network bandwidth and the nature of the future traffic load. However, our model does reveal some interesting phenomena. First, in some circumstances, the amount of incremental bandwidth needed to make a best-effort-only network perform as well as a reservation capable one diverges as capacity increases. Second, in some circumstances reservation-capable networks retain significant advantages over best-effort-only networks, no matter how cheap bandwidth becomes. Lastly, we find bounds on the maximum performance advantage a reservation-capable network can achieve over best effort architectures.", "authors": [ {"name": "Lee Breslau", "email": "breslau@parc.xerox.com", "affiliation": "Xerox Palo Alto Research Center", "contact": true}, {"name": "Scott Shenker", "email": "shenker@parc.xerox.com", "affiliation": "Xerox Palo Alto Research Center"} @@ -269,7 +269,7 @@ { "_id_": 21, "title": "Quality of Service Based Routing: A Performance Perspective", - "abstract": "Recent studies provide evidence that Quality of Service (QoS) routing can provide increased network utilization compared to routing that is not sensitive to QoS requirements of traffic. However, there are still strong concerns about the increased cost of QoS routing, both in terms of more complex and frequent computations and increased routing protocol overhead. The main goals of this application are to study these two cost components, and propose solutions that achieve good routing performance with reduced processing cost. First, we identify the parameters that determine the protocol traffic overhead, namely (a) policy for triggering updates, (b) sensitivity of this policy, and (c) clampdown timers that limit the rate of updates. Using simulation, we study the relative significance of these factors and investigate the relationship between routing performance and the amount of update traffic. In addition, we explore a range of design options to reduce the processing cost of QoS routing algorithms, and study their effect on routing performance. Based on the conclusions of these studies, we develop extensions to the basic QoS routing, that can achieve good routing performance with limited update generation rates. The application also addresses the impact on the results of a number of secondary factors such as topology, high level admission control, and characteristics of network traffic.", + "abstract": "Recent studies provide evidence that Quality of Service (QoS) routing can provide increased network utilization compared to routing that is not sensitive to QoS requirements of traffic. However, there are still strong concerns about the increased cost of QoS routing, both in terms of more complex and frequent computations and increased routing protocol overhead. The main goals of this paper are to study these two cost components, and propose solutions that achieve good routing performance with reduced processing cost. First, we identify the parameters that determine the protocol traffic overhead, namely (a) policy for triggering updates, (b) sensitivity of this policy, and (c) clampdown timers that limit the rate of updates. Using simulation, we study the relative significance of these factors and investigate the relationship between routing performance and the amount of update traffic. In addition, we explore a range of design options to reduce the processing cost of QoS routing algorithms, and study their effect on routing performance. Based on the conclusions of these studies, we develop extensions to the basic QoS routing, that can achieve good routing performance with limited update generation rates. The paper also addresses the impact on the results of a number of secondary factors such as topology, high level admission control, and characteristics of network traffic.", "authors": [ {"name": "George Apostolopoulos", "affiliation": "University of Maryland"}, {"name": "Roch Guérin", "affiliation": "IBM T. J. Watson Research Center"}, @@ -282,7 +282,7 @@ { "_id_": 22, "title": "Scalable QoS Provision Through Buffer Management", - "abstract": "In recent years, a number of link scheduling algorithms have been proposed that greatly improve upon traditional FIFO scheduling in being able to assure rate and delay bounds for individual sessions. However, they cannot be easily deployed in a backbone environ ment with thousands of sessions, as their complexity increases with the number of sessions. In this application, we propose and analyze an approach that uses a simple buffer management scheme to provide rate guarantees to individual flows (or to a set of flows) multiplexed into a common FIFO queue. We establish the buffer allocation re quirements to achieve these rate guarantees and study the trade-off between the achievable link utilization and the buffer size required with the proposed scheme. The aspect of fair access to excess band width is also addressed, and its mapping onto a buffer allocation rule is investigated. Numerical examples are provided that illus trate the performance of the proposed schemes. Finally, a scalable architecture for QoS provisioning is presented that integrates the proposed buffer management scheme with WFQ scheduling that uses a small number of queues.", + "abstract": "In recent years, a number of link scheduling algorithms have been proposed that greatly improve upon traditional FIFO scheduling in being able to assure rate and delay bounds for individual sessions. However, they cannot be easily deployed in a backbone environ ment with thousands of sessions, as their complexity increases with the number of sessions. In this paper, we propose and analyze an approach that uses a simple buffer management scheme to provide rate guarantees to individual flows (or to a set of flows) multiplexed into a common FIFO queue. We establish the buffer allocation re quirements to achieve these rate guarantees and study the trade-off between the achievable link utilization and the buffer size required with the proposed scheme. The aspect of fair access to excess band width is also addressed, and its mapping onto a buffer allocation rule is investigated. Numerical examples are provided that illus trate the performance of the proposed schemes. Finally, a scalable architecture for QoS provisioning is presented that integrates the proposed buffer management scheme with WFQ scheduling that uses a small number of queues.", "authors": [ {"name": "R. Guérin", "email": "guerin@watson.ibm.com", "affiliation": "IBM T. J. Watson Research Center", "contact": true}, {"name": "S. Kamat", "email": "sanjay@watson.ibm.com", "affiliation": "IBM T. J. Watson Research Center", "contact": true}, @@ -295,7 +295,7 @@ { "_id_": 23, "title": "Data networks as cascades: Explaining the multifractal nature of Internet WAN traffic", - "abstract": "In apparent contrast to the well-documented self-similar (i.e., monofractal) scaling behavior of measured LAN traffic, recent studies have suggested that measured TCP/IP and ATM WAN traffic exhibits more complex scaling behavior, consistent with multifractals. To bring multifractals into the realm of networking, this application provides a simple construction based on cascades (also known as multiplicative processes) that is motivated by the protocol hierarchy of IP data networks. The cascade framework allows for a plausible physical explanation of the observed multifractal scaling behavior of data traffic and suggests that the underlying multiplicative structure is a traffic invariant for WAN traffic that co-exists with self-similarity. In particular, cascades allow us to refine the previously observed self-similar nature of data traffic to account for local irregularities in WAN traffic that are typically associated with networking mechanisms operating on small time scales, such as TCP flow control.\n\nTo validate our approach, we show that recent measurements of Internet WAN traffic from both an ISP and a corporate environment are fully consistent with the proposed cascade paradigm and hence with multifractality. We rely on wavelet-based time-scale analysis techniques to visualize and to infer the scaling behavior of the traces, both globally and locally. We also discuss and illustrate with some examples how this cascade-based approach to describing data network traffic suggests novel ways for dealing with networking problems and helps in building intuition and physical understanding about the possible implications of multifractality on issues related to network performance analysis.", + "abstract": "In apparent contrast to the well-documented self-similar (i.e., monofractal) scaling behavior of measured LAN traffic, recent studies have suggested that measured TCP/IP and ATM WAN traffic exhibits more complex scaling behavior, consistent with multifractals. To bring multifractals into the realm of networking, this paper provides a simple construction based on cascades (also known as multiplicative processes) that is motivated by the protocol hierarchy of IP data networks. The cascade framework allows for a plausible physical explanation of the observed multifractal scaling behavior of data traffic and suggests that the underlying multiplicative structure is a traffic invariant for WAN traffic that co-exists with self-similarity. In particular, cascades allow us to refine the previously observed self-similar nature of data traffic to account for local irregularities in WAN traffic that are typically associated with networking mechanisms operating on small time scales, such as TCP flow control.\n\nTo validate our approach, we show that recent measurements of Internet WAN traffic from both an ISP and a corporate environment are fully consistent with the proposed cascade paradigm and hence with multifractality. We rely on wavelet-based time-scale analysis techniques to visualize and to infer the scaling behavior of the traces, both globally and locally. We also discuss and illustrate with some examples how this cascade-based approach to describing data network traffic suggests novel ways for dealing with networking problems and helps in building intuition and physical understanding about the possible implications of multifractality on issues related to network performance analysis.", "authors": [ {"name": "A. Feldmann", "email": "anja@research.att.com", "affiliation": "AT&T Labs—Research", "contact": true}, {"name": "A. C. Gilbert", "email": "agilbert@research.att.com", "affiliation": "AT&T Labs—Research"}, @@ -320,7 +320,7 @@ { "_id_": 25, "title": "Secure Group Communications Using Key Graphs", - "abstract": "Many emerging applications (e.g., teleconference, real-time information services, pay per view, distributed interactive simulation, and collaborative work) are based upon a group communications model, i.e., they require packet delivery from one or more authorized senders to a very large number of authorized receivers. As a result, securing group communications (i.e., providing confidentiality, integrity, and authenticity of messages delivered between group members) will become a critical networking issue. In this application, we present a novel solution to the scalability problem of group/multicast key management. We formalize the notion of a secure group as a triple (U; K; R) where U denotes a set of users, K a set of keys held by the users, and R a user-key relation. We then introduce key graphs to specify secure groups. For a special class of key graphs, we present three strategies for securely distributing rekey messages after a join/leave, and specify protocols for joining and leaving a secure group. The rekeying strategies and join/leave protocols are implemented in a prototype group key server we have built. We present measurement results from experiments and discuss performance comparisons. We show that our group key management service, using any of the three rekeying strategies, is scalable to large groups with frequent joins and leaves. In particular, the average measured processing time per join/leave increases linearly with the logarithm of group size.", + "abstract": "Many emerging applications (e.g., teleconference, real-time information services, pay per view, distributed interactive simulation, and collaborative work) are based upon a group communications model, i.e., they require packet delivery from one or more authorized senders to a very large number of authorized receivers. As a result, securing group communications (i.e., providing confidentiality, integrity, and authenticity of messages delivered between group members) will become a critical networking issue. In this paper, we present a novel solution to the scalability problem of group/multicast key management. We formalize the notion of a secure group as a triple (U; K; R) where U denotes a set of users, K a set of keys held by the users, and R a user-key relation. We then introduce key graphs to specify secure groups. For a special class of key graphs, we present three strategies for securely distributing rekey messages after a join/leave, and specify protocols for joining and leaving a secure group. The rekeying strategies and join/leave protocols are implemented in a prototype group key server we have built. We present measurement results from experiments and discuss performance comparisons. We show that our group key management service, using any of the three rekeying strategies, is scalable to large groups with frequent joins and leaves. In particular, the average measured processing time per join/leave increases linearly with the logarithm of group size.", "authors": [ {"name": "Chung Kei Wong", "email": "ckwong@cs.utexas.edu", "affiliation": "Department of Computer Sciences, University of Texas at Austin", "contact": true}, {"name": "Mohamed Gouda", "email": "gouda@cs.utexas.edu", "affiliation": "Department of Computer Sciences, University of Texas at Austin"}, @@ -332,7 +332,7 @@ { "_id_": 26, "title": "Achieving bounded fairness for multicast and TCP traffic in the internet", - "abstract": "There is an urgent need for effective multicast congestion control algorithms which enable reasonably fair sharing of network resources between multicast and unicast TCP traffic under the current Internet infrastructure. In this application, we propose a quantitative definition of a type of bounded fairness between multicast and unicast best-effort traffic, termed essentially fair. We also propose a window-based Random Listening Algorithm (RLA) for multicast congestion control. The algorithm is proven to be essentially fair to TCP connections under a restricted topology with equal round-trip times and with phase effects eliminated. The algorithm is also fair to multiple multicast sessions. This application provides the theoretical proofs and some simulation results to demonstrate that the RLA achieves good performance under various network topologies. These include the performance of a generalization of the RLA algorithm for topologies with different round-trip times.", + "abstract": "There is an urgent need for effective multicast congestion control algorithms which enable reasonably fair sharing of network resources between multicast and unicast TCP traffic under the current Internet infrastructure. In this paper, we propose a quantitative definition of a type of bounded fairness between multicast and unicast best-effort traffic, termed essentially fair. We also propose a window-based Random Listening Algorithm (RLA) for multicast congestion control. The algorithm is proven to be essentially fair to TCP connections under a restricted topology with equal round-trip times and with phase effects eliminated. The algorithm is also fair to multiple multicast sessions. This paper provides the theoretical proofs and some simulation results to demonstrate that the RLA achieves good performance under various network topologies. These include the performance of a generalization of the RLA algorithm for topologies with different round-trip times.", "authors": [ {"name": "Huayan Amy Wang", "email": "whycu@ctr.columbia.edu", "affiliation": "Department of Electrical Engineering, Columbia University", "contact": true}, {"name": "Mischa Schwartz", "email": "schwartz@ctr.columbia.edu", "affiliation": "Department of Electrical Engineering, Columbia University"} @@ -358,7 +358,7 @@ { "_id_": 28, "title": "Session Directories and Internet Multicast Address Allocation", - "abstract": "A multicast session directory is a mechanism by which users can discover the existence of multicast sessions. In the Mbone, session announcements have also served as multicast address reservations - a dual purpose that is efficient, but which may cause some side-affects as session directories scale.\n\nIn this application we examine the scaling of multicast address allocation when it is performed by such a multicast session directory. Despite our best efforts to make such an approach scale, this analysis ultimately reveals significant scaling problems, and suggests a new approach to multicast address allocation in the Internet environment.", + "abstract": "A multicast session directory is a mechanism by which users can discover the existence of multicast sessions. In the Mbone, session announcements have also served as multicast address reservations - a dual purpose that is efficient, but which may cause some side-affects as session directories scale.\n\nIn this paper we examine the scaling of multicast address allocation when it is performed by such a multicast session directory. Despite our best efforts to make such an approach scale, this analysis ultimately reveals significant scaling problems, and suggests a new approach to multicast address allocation in the Internet environment.", "authors": [ {"name": "Mark Handley", "email": "mjh@isi.edu", "affiliation": "USC Information Sciences Institute", "contact": true} ], @@ -368,7 +368,7 @@ { "_id_": 29, "title": "*Core*-Stateless Fair Queueing: Achieving Approximately Fair Bandwidth Allocations in High Speed Networks", - "abstract": "Router mechanisms designed to achieve fair bandwidth allocations, like Fair Queueing, have many desirable properties for congestion control in the Internet. However, such mechanisms usually need to maintain state, manage buffers, and/or perform packet scheduling on a per ow basis, and this complexity may prevent them from being cost-effectively implemented and widely deployed. In this application, we propose an architecture that significantly reduces this implementation complexity yet still achieves approximately fair bandwidth allocations. We apply this approach to an island of routers { that is, a contiguous region of the network { and we distinguish between edge routers and core routers. Edge routers maintain per ow state; they estimate the incoming rate of each ow and insert a label into each packet header based on this estimate. Core routers maintain no per ow state; they use FIFO packet scheduling augmented by a probabilistic dropping algorithm that uses the packet labels and an estimate of the aggregate traffic at the router. We call the scheme Core-Stateless Fair Queueing. We present simulations and analysis on the performance of this approach, and discuss an alternate approach.", + "abstract": "Router mechanisms designed to achieve fair bandwidth allocations, like Fair Queueing, have many desirable properties for congestion control in the Internet. However, such mechanisms usually need to maintain state, manage buffers, and/or perform packet scheduling on a per ow basis, and this complexity may prevent them from being cost-effectively implemented and widely deployed. In this paper, we propose an architecture that significantly reduces this implementation complexity yet still achieves approximately fair bandwidth allocations. We apply this approach to an island of routers { that is, a contiguous region of the network { and we distinguish between edge routers and core routers. Edge routers maintain per ow state; they estimate the incoming rate of each ow and insert a label into each packet header based on this estimate. Core routers maintain no per ow state; they use FIFO packet scheduling augmented by a probabilistic dropping algorithm that uses the packet labels and an estimate of the aggregate traffic at the router. We call the scheme Core-Stateless Fair Queueing. We present simulations and analysis on the performance of this approach, and discuss an alternate approach.", "authors": [ {"name": "Ion Stoica", "email": "istoica@cs.cmu.edu", "affiliation": "CMU", "contact": true}, {"name": "Scott Shenker", "email": "shenker@parc.xerox.com", "affiliation": "Xerox PARC"}, @@ -380,7 +380,7 @@ { "_id_": 30, "title": "Uniform versus Priority Dropping for Layered Video", - "abstract": "In this application, we analyze the relative merits of uniform versus priority dropping for the transmission of layered video. We first present our original intuitions about these two approaches, and then investigate the issue more thoroughly through simulations and analysis in which we explicitly model the performance of layered video applications. We compare both their performance characteristics and incentive properties, and find that the performance benefit of priority dropping is smaller than we expected, while uniform dropping has worse incentive properties than we previously believed.", + "abstract": "In this paper, we analyze the relative merits of uniform versus priority dropping for the transmission of layered video. We first present our original intuitions about these two approaches, and then investigate the issue more thoroughly through simulations and analysis in which we explicitly model the performance of layered video applications. We compare both their performance characteristics and incentive properties, and find that the performance benefit of priority dropping is smaller than we expected, while uniform dropping has worse incentive properties than we previously believed.", "authors": [ {"name": "Sandeep Bajaj", "email": "bajaj@parc.xerox.com", "affiliation": "Xerox Palo Alto Research Center"}, {"name": "Lee Breslau", "email": "breslau@parc.xerox.com", "affiliation": "Xerox Palo Alto Research Center"}, diff --git a/test/emails.txt b/test/emails.txt index 47b797c98..91b7e6070 100644 --- a/test/emails.txt +++ b/test/emails.txt @@ -1922,7 +1922,7 @@ conference. Chris Lilley (World Wide Web Consortium) * Site: http://hotcrp.lcdf.org/test/paper/14?cap={{}} -0 applications were accepted out of {{}} submissions. +0 papers were accepted out of {{}} submissions. Visit the submission site for reviews, comments, and related information. Reviews and comments are also included below. @@ -1943,7 +1943,7 @@ Reviewer expertise ------------------ 2. Some familiarity -Application summary +Paper summary ------------- Summary V @@ -1973,7 +1973,7 @@ conference. Chris Lilley (World Wide Web Consortium) * Site: http://hotcrp.lcdf.org/test/paper/14?cap={{}} -0 applications were accepted out of {{}} submissions. +0 papers were accepted out of {{}} submissions. Visit the submission site for reviews, comments, and related information. Reviews and comments are also included below.