[pilc] A few comments on LINK version 11
Spencer Dawkins <spencer_dawkins@yahoo.com> Tue, 11 June 2002 18:06 UTC
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Subject: [pilc] A few comments on LINK version 11
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Dear All, I've been through LINK again. When PILC was formed, the expectation was that this document would be our primary output, with the (aready-approved) BCPs helping to bandaid TCP over already-deployed subnetwork technologies. I believe that Phil and Mark have met this expectation, with help from a number of others. I do have a few comments on version 11... Spencer ------------ 1 Introduction and Overview I think the following is too polite! Because IP is so simple, consideration 2 is more of an issue than consideration 1. I.e., subnetwork designers make many more errors of commission than errors of omission. But certain enhanced Internet features, such as multicasting and quality-of-service, benefit significantly from support from the underlying subnetworks beyond that necessary to carry "traditional" unicast, best-effort IP. I propose IP transport requirements are so minimal that consideration 2 is more of an issue than consideration 1. Subnetwork designers make many more errors of commission than errors of omission. o A frequent error of commission is designing a subnetwork that provides a high level of subnetwork-level reliability for IP datagrams, at a corresponding high cost in latency and jitter - even though higher-level protocols can't assume this level of reliability for all IP paths and must duplicate this reliability if it's needed. o When subnetwork designers do make errors of omission, it's usually by not providing subnetwork-level support for multicast transport and quality-of-service mechanisms - or anything else beyond "traditional" unicast, best-effort IP transport. Would it make sense to point to RFC 3135 at the end of the following paragraph? However, partial duplication of functionality in a lower layer can *sometimes* be justified by performance, security or availability considerations. Examples include link-layer retransmission to improve the performance of an unusually lossy channel, e.g., mobile radio; link level encryption intended to thwart traffic analysis; and redundant transmission links to improve availability, increase throughput, or to guarantee performance for certain classes of traffic. Duplication of protocol function should be done only with an understanding of system level implications, including possible interactions with higher-layer mechanisms. 2 Maximum Transmission Units (MTUs) and IP Fragmentation I had too may "it is" in this paragraph to parse quickly: If optional header compression [RFC1144] [RFC2507] [RFC2508] [RFC3095] is used, however, it is required that the link framing indicate frame length as it is needed for the reconstruction of the original header. I propose that we use more nouns, as follows: If optional header compression [RFC1144] [RFC2507] [RFC2508] [RFC3095] is used, however, link framing mechanisms are required to indicate frame length as this information is needed to reconstruct the original header. Do we still think P-MTU black holes are a sufficent problem that we will still include this problem on the following list of justifications for support of large MTU sizes? (If so, should P-MTU discovery still be on the 2.5G3G list of recommendations? But that's not a LINK question...) The Path MTU Discovery procedure, the persistence of path MTU black holes, and the deletion of router fragmentation in IPv6 reflects a consensus of the Internet technical community that router fragmentation is best avoided. This requires that subnetworks support MTUs that are "reasonably" large. The smallest MTU permitted in IPv4 by [RFC791] is 68 bytes, but such a small value would clearly be inefficient. Because IPv6 omits fragmentation by routers, [RFC 2460] specifies a larger minimum MTU of 1280 bytes. Any subnetwork with an internal packet payload smaller than 1280 bytes must implement a mechanism that performs fragmentation/reassembly of IP packets to/from subnetwork frames if it is to support IPv6. Is there an interaction here between internal fragmentation and insertion of a smaller/higher priority packet during transmission of a larger/lower priority packet, for reasons described in 2.1, and using the multiplexing mechanism described in 2.1? If a subnetwork cannot directly support a "reasonable" MTU with native framing mechanisms, it should internally fragment. That is, it should transparently break IP packets into internal data elements and reassemble them at the other end of the subnetwork. This leaves the question of what is a "reasonable" MTU. Ethernet (10 and 100 Mb/s) has a MTU of 1500 bytes, and because of its ubiquity Isn't 1500 bytes the limit for all 802.3s (including 1 and 10 Gb/s)? few Internet paths have MTUs larger than this value. This severely limits the utility of larger MTUs provided by other subnetworks. Meanwhile larger MTUs are increasingly desirable on high speed subnetworks to reduce the per-packet processing overhead in host computers, and implementers are encouraged to provide them even though they may not be usable when Ethernet is also in the path. Various "tunneling" schemes, such as IP Security [RFC2406] treat IP as a subnetwork for IP. Since tunneling adds header overhead, it can Doesn't insertion of MPLS headers have the same effect? trigger fragmentation even when the same physical subnetworks (e.g., Ethernet) are used on both sides of the IP router. Tunneling has made it more difficult to avoid router fragmentation and has increased the incidence of path MTU black holes. Larger subnetwork MTUs may help to alleviate this problem. I think it's also reasonable to point out that P-MTU discovery uses a table of "likely" MTU sizes that doesn't map very well to what used to be full-sized MSSes that are being tunneled - the fallback to the first working MTU in the table may be quite a bit smaller than the un-tunneled P-MTU... I wonder if this table will ever be updated... 2.1 Choosing the MTU in Slow Networks In slow networks, the largest possible packet may take a considerable time to send. Interactive response time should not exceed the well- Can (should) we say "Round-trip times for interactive responses should not exceed"? Also, can we point to RFC 1144 as a source for the 100-200 ms requirement for interactive response? known human factors limit of 100 to 200 ms. This includes all sources of delay: electromagnetic propagation delay, queuing delay, and the store-and-forward time, i.e,. the time to transmit a packet at link speed. 3 Framing on Connection-Oriented Subnetworks (no comments in this section) 4 Connection-Oriented Subnetworks Because Internet traffic is typically bursty and transaction- oriented, it is often difficult to pick an optimal idle timeout. If the timeout is too short, subnetwork connections are opened and closed rapidly, possibly over-stressing its call management system (especially if was designed for voice traffic holding times). If the timeout is too long, subnetwork connections are idle much of the time, wasting any resources dedicated to them by the subnetwork. Can we also add the following: Subscribers may also be charged differently depending on whether a subnetwork leaves idle connections in place using a relatively-long idle timer or tears down idle connections with little delay and reestablishes these connections after a short period of time. If it makes sense to do so, subscribers should be given control over this behavior (for example, PPP implementations might allow subscribers to control an idle timer for locally- initiated dialup connections. Should be "is used TO support IP" in the following sentence. In any event, if an SNDCF that opens and closes subnet connections is used support IP, care should be taken to make sure that call processing in the subnet can keep up with relatively short holding times. 5 Broadcasting and Discovery Subnetworks fall into two categories: point-to-point and shared. A point-to-point subnet has exactly two endpoint components (hosts or routers); a shared link has more than two, using either an inherent broadcast media (e.g., Ethernet, radio) or on a switching layer hidden from the network layer (switched Ethernet, Myrinet [MYR], ATM). Switched subnetworks handle broadcast by copying broadcast Could this sentence say "by forwarding broadcast packets on all other interfaces to ensure..."? packets to ensure each end system receives a copy of each packet. Could we insert "Centralized databases also insert new failure points and scaling hot-spots into the network." before the last sentence in the following paragraph? The lack of broadcast can impede the performance of these protocols, or render them inoperable (e.g. DHCP). ARP-like link address lookup can be provided by a centralized database, but at the expense of potentially higher response latency and the need for nodes to have explicit knowledge of the ARP server address. Shared links should support native, link-layer subnet broadcast. 6 Multicasting Receivers also need to be designed to accept packets addressed to some number of multicast addresses in addition to the unicast packets specifically addressed to them. How many multicast addresses need to be supported by a host depends on the requirements of the associated host; at least several dozen will meet most current needs. The last phrase doesn't parse well for me. Is it saying "few hosts must accept multicast packets from more than a few dozen multicast addresses"? On low-speed networks the multicast address recognition function may be readily implemented in host software, but on high speed networks it should be implemented in subnetwork hardware. This hardware need not be complete; for example, many Ethernet interfaces implement a "hashing" function that passes all of the multicast (and unicast) traffic to which the associated host subscribes, plus some small fraction of multicast traffic to which the host does not subscribe. Host/router software then has to discard the unwanted packets that pass the hardware filter. There does not need to be a one-to-one mapping between subnetwork multicast address and IP multicast address. An address overlap may significantly degrade the filtering capability of a receiver's hardware multicast address filter. A subnetwork supporting only Is this sentence saying "If more than one IP multicast address makes into a subnetwork multicast address, this many-to-one mapping may significantly degrade the filtering capacity of a receiver's hardware multicast address filter", or something else? "Address overlap" is too ambiguous for me... broadcast should use this service for multicast and must rely on software filtering. 7 Bandwidth on Demand (BoD) Subnets Long delay BoD subnets pose problems similar to connection oriented networks in anticipating traffic. While connection oriented subnets hold idle channels open expecting new data to arrive, BoD subnets request channel access based on buffer occupancy (or expected buffer occupancy) on the sending port. Poor performance will likely result if the sender does not anticipate additional traffic arriving at that I'm confused here. Is this sentence saying "Poor performance will likely resuly if the sending host does not queue additional outgoing traffic while the sender is waiting for its transmission request to be granted"? If so, I agree. If not, can we talk? I may not disagree, I just don't understand the point being made. port during the time it takes to grant a transmission request. It is recommended that the algorithm have the capability to extend a hold on the channel for data that has arrived after the original request was generated (this may done by piggybacking new requests on user data). There are a wide variety of BoD protocols available. However, there has been relatively little comprehensive research on the interactions between the BoD mechanisms and Internet protocol performance. Research on some specific mechanisms is available (e.g., [AR02]). One item that has been studied is TCP's retransmission timer [KY02]. BoD systems can cause spurious timeouts when adjusting from a relatively high data rate to a relatively low data rate. In this case, TCP's transmitted data takes longer to get through the network than predicted by the retransmission timeout (RTO) and therefore the TCP sender is prone to resending a segment prematurely. Could we also include the observation that delays to acquire a shared control channel may also be reflected as sudden and unpredictable "spikes" in apparent round-trip times, making RTO values "spike" unpredictably as well? I'm trying to say that it's not just the change in available bandwidth, but delays encountered while changing available bandwidth as well... 8 Reliability and Error Control In the Internet architecture, the ultimate responsibility for error recovery is at the end points. The Internet may occasionally drop, corrupt, duplicate or reorder packets, and the transport protocol (e.g., TCP) or application (e.g., if UDP is used as the transport protocol) must recover from these errors on an end-to-end basis. I'm not sure this is strictly true - I'm thinking of RTP over UDP, where the RTP sender may use RTCP indications of loss to lower its sending rates. Is this "recovering" in the "reliability and error control" sense? I think it's not "must recover", but "recovers if necessary". Error recovery in the subnetwork is therefore justified only to the extent that it can enhance overall performance. It is important to recognize that a subnetwork can go too far in attempting to provide error recovery services in the Internet environment. Subnet reliability should be "lightweight", i.e., it only has to be "good enough", *not* perfect. Can we add something like: Applications have no way of accepting a lower level of reliability (perhaps as a tradeoff for lower latency) than the subnetwork provides. "Reliability" is a good thing until it prevents users from using an application over a too-reliable subnetwork. In this section we discuss how to analyze characteristics of a subnetwork to determine what is "good enough". The discussion below focuses on TCP, which is the most widely used transport protocol in the Internet. It is widely believed (and is a stated goal within the IETF) that non-TCP transport protocols should attempt to be "TCP- friendly" and have many of the same performance characteristics. Is this true? I thought the point of SCTP and DCP was have "safe" behavior when facing congestion, but to vary other dimensions of performance. Thus, the discussion below should be applicable even to portions of the Internet where TCP may not be the predominant protocol. I think the discussion is applicable, but not just because all transport protocols have the same performance characteristics as TCP. 8.1 TCP vs Link-Layer Retransmission The use of ECC to detect transmission errors so that retransmissions (hopefully without errors) can be requested is widely known as "ARQ" (Automatic Repeat Request). Can we make it "Automatic Repeat reQuest", to match the acronym? This inter-layer "competition" might lead to the following wasteful situation. When the link layer retransmits (parts of) a packet, the link latency momentarily increases. Since TCP bases its retransmission timeout on prior measurements of end-to-end latency, including that of the link in question, this sudden increase in latency may trigger an unnecessary retransmission by TCP of a packet that the link layer is still retransmitting. Such spurious end-to- end retransmissions generate unnecessary load and reduce end-to-end throughput. One may even have multiple copies of the same packet in the same link queue at the same time. In general, one could say the competing error recovery is caused by an inner control loop (link- layer error recovery) reacting to the same signal as an outer control loop (end- to-end error recovery) without any coordination between the loops. Note that this is solely an efficiency issue; TCP continues to provide reliable end-to-end delivery over such links. Can we append ", although this delivery is being slowed by unnecessary link-layer transmissions"? 8.2 Recovery from Subnetwork Outages Under these circumstances TCP will, after each unsuccessful retransmission, wait even longer before trying again; this is its "exponential back-off" algorithm. Furthermore, TCP will not discover that the subnetwork outage has ended until its next retransmission attempt. If TCP has backed off, this may take some time. This can Can we say "this may take a number of minutes."? I'm still talking to people who find these long timer backoffs surprising. lead to extremely poor TCP performance over such subnetworks. It is therefore highly desirable that a subnetwork subject to outages not silently discard packets during an outage. Ideally, it should define an interface to the next higher layer (i.e., IP) that allows it to refuse packets during an outage, and to automatically ask IP for new packets when it is again able to deliver them. If it cannot do this, then the subnetwork should hold onto at least some of the We're saying "at least some" in this paragraph and "only a single packet per TCP connection, including ACKs" in the next paragraph. Can we make one recommendation? packets it accepts during an outage and attempt to deliver them when the subnetwork comes back up. When packets are discarded, IP should This is "When outbound packets are discarded,", right? be notified so that the appropriate ICMP messages can be sent. Only a single packet per TCP connection, including ACKs, need be held in this way to cause the TCP sender to recover from the additional losses once the flow resumes [ARQ-DRAFT]. (See previous note about the right number of packets to hang onto during outages.) Is there a maximum period of time that a subnetwork should hang onto a packet? I don't feel good about "longer than 2 * MSL" - would sending packets that are older than this cause problems? 8.3 CRCs, Checksums and Error Detection The TCP [RFC793], UDP [RFC768], ICMP, and IPv4 [RFC791] protocols all use the same simple 16-bit 1's complement checksum algorithm to detect corrupted packets. The IPv4 checksum protects only the IPv4 header, while the TCP, ICMP, and UDP checksums provide end-to-end error detection for both the transport pseudo header (including network and transport layer information) and the transport payload data. Protection of the data is optional for applications using UDP [RFC768]. Do we think UDP checksums are "optional but recommended", based on the NFS war stories, or is this going too far? My understanding is that UDP checksums are enabled host-wide on most implementations - would this make turning UDP checksums off less desirable, because errored packets might be delivered to any running application on the host? One way to provide additional protection for an IPv4 or IPv6 header is by the authentication and packet integrity services of the IP Security (IPSEC) protocol [RFC2401]. However, this may not be a choice available to the subnetwork designer. Is this just saying IPSEC AH is an IP mechanism, or is it saying more? 8.4 How TCP Works TCP uses sequence numbering and acknowledgments (ACKs) on an end-to- end basis to provide reliable, sequenced, once-only delivery. TCP ACKs are cumulative, i.e., each implicitly ACKs every segment received so far. If an ACK is not received, the acknowledgment value Isn't this "implicitly ACKs all previously unacknowledged data in the sender's outgoing window"? carried in the cumulative packet will cease to advance. Since the most common cause of packet loss is congestion, TCP treats packet loss as an Internet congestion indicator. This happens automatically, and the subnetwork need not know anything about IP or TCP. It simply drops packets whenever it must, though some packet- dropping strategies (e.g., RED) are more fair to competing flows than others. Is this a fairness issue? I might have said "reduce the effect of congestion on competing flows, compared to others (tail drop)." The following seems to ignore slow start: TCP recovers from packet losses in two different ways. The most important is the retransmission timeout. If an ACK fails to arrive after a certain period of time, TCP retransmits the oldest unacked packet. Taking this as a hint that the network is congested, TCP waits for the retransmission to be ACKed before it continues, and it gradually increases the number of packets in flight as long as a timeout does not occur again. Restart behavior is more correctly described in the next paragraph - can we drop the last clause of the previous paragraph ("... and it gradually ...")? Then we go straight into: A retransmission timeout can impose a significant performance penalty, as the sender is idle during the timeout interval and restarts with a congestion window of 1 following the timeout. To allow faster recovery from the occasional lost packet in a bulk transfer, an alternate scheme known as "fast recovery" was introduced [RFC2581] [RFC2582] [RFC2914] [TCPF98]. This discussion of fast recovery seems so loose that it's actually about fast retransmit: Fast recovery relies on the fact that when a single packet is lost in "Fast retransmit relies on the fact that even when one or more oackets are lost in a bulk transfer, the receiver may continue to return ACKs to ..." a bulk transfer, the receiver continues to return ACKs to subsequent data packets that do not actually acknowledge any newly-received data. These are known as "duplicate acknowledgments" or "dupacks". The sending TCP can use dupacks as a hint that a packet has been lost and retransmit it without waiting for a timeout. Dupacks effectively constitute a negative acknowledgment (NAK) for the packet sequence number in the acknowledgment field. TCP waits until a certain number of dupacks (currently 3) are seen prior to assuming a loss has occurred; this helps avoid an unnecessary retransmission during out- of-sequence delivery. A new technique called "Explicit Congestion Notification" (ECN) [RFC3168] allows routers to directly signal congestion to hosts without dropping packets. This is done by setting a bit in the IP header. Since this is currently an optional behavior (and, longer term, there will always be the possibility of congestion in portions of the network which don't support ECN), the lack of an ECN bit must NEVER be interpreted as a lack of congestion. Thus, for the "lack of congestion - especially since ECN-capable routers still drop packets at high levels of congestion, instead of continuing to send traffic on congested links. Thus, fot the foreseeable future ..." foreseeable future, TCP must interpret a lost packet as a signal of congestion. The TCP "congestion avoidance" [RFC2581] algorithm maintains a congestion window (cwnd) controlling the amount of data TCP may have in flight at any moment. Reducing cwnd reduces the overall bandwidth obtained by the connection; similarly, raising cwnd increases the performance, up to the limit of the available bandwidth. "reduces the overall bandwidth that the sender will attempt to use for the connection; similarly ..." TCP probes for available network bandwidth by setting cwnd to one packet and then increasing it by one packet for each ACK returned from the receiver. This is TCP's "slow start" mechanism. When a packet loss is detected (or congestion is signaled by other mechanisms), cwnd is reset to one and the slow start process is repeated until cwnd reaches one half of its previous setting before the reset. "This is the "slowstart threshold", or ssthresh." Cwnd continues to increase past this point, but at a much slower rate than before. If no further losses occur, cwnd will ultimately reach the window size advertised by the receiver. 8.5 TCP Performance Characteristics No comments. 8.5.1 The Formulae No comments. 8.5.2 Assumptions Both of these formulae allow BW to become infinite if there is no loss. This is because an Internet path will drop packets at bottleneck queues if the load is too high. Thus, a completely lossless TCP/IP network can never occur (unless the network is being underutilized). I don't understand the last sentence in the previous paragraph. Is it saying "A completely lossless TCP/IP connection can never occur, unless the receiver is using a receive window smaller than the bandwidth-delay product"? 8.5.3 Analysis of Link-Layer Effects on TCP Performance No comments. 9 Quality-of-Service (QoS) considerations No comments. 10 Fairness vs Performance No comments. 11 Delay Characteristics No comments. 12 Bandwidth Asymmetries Some subnetworks may provide asymmetric bandwidth (or may cause TCP packet flows to experience asymmetry in the capacity) and the Internet protocol suite will generally still work fine. However, there is a case when such a scenario reduces TCP performance. Since TCP data segments are 'clocked' out by returning acknowledgments TCP senders are limited by the rate at which ACKs can be returned [BPK98]. Therefore, when the ratio of the bandwidth of the subnetwork carrying the data to the bandwidth of the subnetwork carrying the acknowledgments is too large, the slow return of of the ACKs directly impacts performance. Since ACKs are generally smaller than data segments, TCP can tolerate some asymmetry, but as a general Should we say "TCPs can tolerate substantial assymetry, ..."? rule designers of subnetworks should be aware that subnetworks with significant asymmetry can result in reduced performance, unless issues are taken to mitigate this [asym-id]. Several strategies have been identified for reducing the impact of asymmetry of the network path between two TCP end hosts, e.g. [asym- id]. These techniques attempt to reduce the number of ACKs transmitted over the return path (low bandwidth channel) by changes at the end host(s), and/or by modification of subnetwork packet forwarding. While these solutions may mitigate the performance issues caused by asymmetric subnetworks, they do have associated cost and may have other implications. A fuller discussion of strategies and their implications is provided in [asym-id]. The IESG documents under review is showing Allison holding this token. 13 Buffering, flow & congestion control No comments. 14 Compression We make a stronger recommendation that subnetworks operating at low speed or with small MTUs compress IP and transport-level headers (TCP and UDP) using several header compression schemes developed within the IETF. An uncompressed 40-byte TCP/IP header takes about 33 milliseconds to send at 9600 bps. "VJ" TCP/IP header compression Can we introduce this as "Van Jacobson TCP/IP header compression", to be consistent with what we settled on in 2.5G/3G? [RFC1144] compresses most headers to 3-5 bytes, reducing transmission time to several milliseconds. This is especially beneficial for small, latency-sensitive packets in interactive sessions. 15 Packet Reordering No comments. 16 Mobility No comments. 17 Routing No comments. 18 Security Considerations Too many transitions in this paragraph? Nonetheless, end-to-end security mechanisms are not used as widely as might be desired. However, the group could not reach consensus on whether subnetwork designers should be actively encouraged to implement mechanisms to protect user data. I propose: Although end-to-end security mechanisms are not used as widely as might be desired, the PILC working group could not reach consensus on whether subnetwork designers should be actively encouraged to implement mechanisms to protect user data. And can we drop the "however, " in this paragraph? It doesn't seem to be in contrast to anything... However, traffic analysis is a notoriously subtle and difficult threat to understand and defeat, far more so than threats to confidentiality and integrity. We therefore urge extreme care in the design of subnetwork security mechanisms specifically intended to thwart traffic analysis. Not to be pedantic, but [WFBA2000] isn't about buffer overruns in I/O system calls - it's about buffer overruns in the standard C library: In addition, well-designed security protocols can be compromised by implementation defects. Examples of such defects include use of predictable pseudo-random numbers [RFC1750], vulnerability to buffer overflow attacks due to unsafe use of certain I/O system calls [WFBA2000], and inadvertent exposure of secret data. References [AP99] M. Allman, V. Paxson, On Estimating End-to-End Network Path Properties, In Proceedings of ACM SIGCOMM 99. A URL for this paper is: http://www.acm.org/sigcomm/sigcomm99/papers/session7-3.html [AR02] G. A,car and C. Rosenberg, Weighted Fair Bandwidth-on-Demand (WFBoD) for Geo-Stationary Satellite Networks with On-Board Processing, Special Issue on Broadband Satellite Systems: A Network Perspective, Computer Networks, accepted on July 13, 2001. To appear in 2002. The best URL I could find for this paper was http://www.iis.ee.ic.ac.uk/~g.acar/Research.html#WFBoD [asym-id] H. Balakrishnan, V. N. Padmanabhan, G. Fairhurst, M, Sooriyabandara. "TCP Performance Implications of Network Path Asymmetry", work in progress as internet-draft, draft-ietf-pilc- asym-07.txt, November 2001. This draft should have already expired. Aaron - are we going to progress it? [ARQ-DRAFT] Fairhurst, G., and L. Wood, Advice to link designers on link Automatic Repeat reQuest (ARQ), work in progress as internet- draft, draft-ietf-pilc-link-arq-issues-03.txt, August 2001. This is now version 04, from March 2002. [ATMFTM] The ATM Forum, "Traffic Management Specification, Version 4.0", April 1996, document af-tm-0056.000 (www.atmforum.com). A more complete URL is ftp://ftp.atmforum.com/pub/approved-specs/af-tm-0056.000.pdf [BA02] Ethan Blanton, Mark Allman. On Making TCP More Robust to Packet Reordering. ACM Computer Communication Review, 32(1), January 2002. A complete URL is http://www.acm.org/sigcomm/ccr/archive/2002/jan02/ccr-200201-allman.pdf [BPK98] Hari Balakrishnan, Venkata Padmanabhan, Randy H. Katz. 'The Effects of Asymmetry on TCP Performance." ACM Mobile Networks and Applications (MONET), 1998. A complete URL is http://daedalus.cs.berkeley.edu/publications/tcpasym-mobicom97.ps.gz [Crypto9912] Schneier, Bruce "European Cellular Encryption Algorithms" Crypto-Gram (December 15, 1999) www.counterpane.com A complete URL is http://www.counterpane.com/crypto-gram-9912.html [DOCSIS3] W.S. Lai, "DOCSIS-Based Cable Networks: Impact of Large Data Packets on Upstream Capacity", 14th ITC Specialists Seminar on Access Networks and Systems, Barcelona, Spain, April 25-27, 2001. A complete URL is http://www.att.com/networkandperformance/docs/ndrp_itc0401b.doc [ES00] David A. Eckhardt and Peter Steenkiste, "Effort-limited Fair (ELF) Scheduling for Wireless Networks, Proceedings of IEE Infocom 2000. The closing quote mark is missing (after "Networks"). A complete URL is http://www.ieee-infocom.org/2000/papers/266.ps [FB00] Firoiu V., and Borden M., "A Study of Active Queue Management for Congestion Control" to appear in Infocom 2000 The paper has already appeared. The URL is http://www.ieee-infocom.org/2000/papers/405.pdf [IEEE80211] IEEE 802.11 Wireless LAN standard. Available from http://standards.ieee.org/catalog/IEEE802.11.html. This is now freely downloadable from http://standards.ieee.org/getieee802/download/802.11-1999.pdf [LKJK02] R. Ludwig, A. Konrad, A. D. Joseph, R. H. Katz, "Optimizing the End-to-End Performance of Reliable Flows over Wireless Links", Kluwer/ACM Wireless Networks Journal, Vol. 8, Nos. 2/3, pp. 289-299, March-May 2002. A Complete URL is http://iceberg.cs.berkeley.edu/papers/Ludwig-Mobicom99/ [LRKOJ99] R. Ludwig, B. Rathonyi, A. Konrad, K. Oden, A. Joseph, Multi-Layer Tracing of TCP over a Reliable Wireless Link, pp. 144-154, In Proceedings of ACM SIGMETRICS 99. Need quotes around title. A complete URL is http://iceberg.cs.berkeley.edu/papers/Ludwig-Sigmetrics99/ [LS00] R. Ludwig, K. Sklower, The Eifel Retransmission Timer, ACM Computer Communication Review, Vol. 30, No. 3, July 2000. Need quotes around title. A complete URL is http://www.acm.org/sigcomm/ccr/archive/2000/july00/ccr_200007-ludwig.html [MBB00] May, M., Bonald, T., and Bolot, J-C., "Analytic Evaluation of RED Performance" to appear INFOCOM 2000 The paper has appeared. A complete URL is http://www.ieee-infocom.org/2000/papers/369.ps [MBDL99] May, M., Bolot, J., Diot, C., and Lyles, B., "Reasons not to deploy RED", technical report, June 1999. A complete URL is http://www-sop.inria.fr/rodeo/personnel/mmay/may_red.html [GM02] Luigi Alfredo Grieco1, Saverio Mascolo, "TCP Westwood and Easy RED to Improve Fairness in High-Speed Networks", Proceedings of the 7th International Workshop on Protocols for High-Speed Networks, April 2002. A complete URL is http://www.cs.ucla.edu/NRL/hpi/papers/2002-pfhsn-0.pdf.gz [MSMO97] M. Mathis, J. Semke, J. Mahdavi, T. Ott, "The Macroscopic Behavior of the TCP Congestion Avoidance Algorithm", Computer Communication Review, volume 27, number 3, July 1997. A complete URL is http://www.psc.edu/networking/papers/model_abstract.html [PFTK98] Padhye, J., Firoiu, V., Towsley, D., and Kurose, J., "Modeling TCP Throughput: a Simple Model and its Empirical Validation", UMASS CMPSCI Tech Report TR98-008, Feb. 1998. A complete URL is ftp://gaia.cs.umass.edu/pub/Padhye-Firoiu98-TCP-throughput-TR.ps [RF95] Romanow, A., and Floyd, S., "Dynamics of TCP Traffic over ATM Networks". IEEE JSAC, V. 13 N. 4, May 1995, p. 633-641. A complete URL is http://www-nrg.ee.lbl.gov/papers/tcp_atm.pdf [RFC1812] Needs details! [Schneier3] Schneier, Bruce "Why Cryptography is Harder Than it Looks", www.counterpane.com A complete URL is http://www.counterpane.com/whycrypto.html [TCPF98] Dong Lin and H.T. Kung, "TCP Fast Recovery Strategies: Analysis and Improvements", IEEE Infocom, March 1998. Available from: "http://www.eecs.harvard.edu/networking/papers/infocom-tcp- final-198.pdf" This URL is broken: [WFBA2000] David Wagner, Jeffrey S. Foster, Eric Brewer and Alexander Aiken, "A First Step Toward Automated Detection of Buffer Overrun Vulnerabilities", Proceedings of NDSS2000, or www.berkeley.edu:80/~daw/papers/ A correct URL is http://www.cs.berkeley.edu/~daw/papers/overruns-ndss00.ps __________________________________________________ Do You Yahoo!? Yahoo! - Official partner of 2002 FIFA World Cup http://fifaworldcup.yahoo.com _______________________________________________ pilc mailing list pilc@ietf.org https://www1.ietf.org/mailman/listinfo/pilc http://www.ietf.org/html.charters/pilc-charter.html http://pilc.grc.nasa.gov/
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