Internet Engineering Task Force Sally Floyd, Editor INTERNET DRAFT ACIRI draft-floyd-cong-00.txt October 1999 Expires: April 2000 Congestion Control Principles Status of this Memo This document is an Internet-Draft and is in full conformance with all provisions of Section 10 of RFC2026. Internet-Drafts are working documents of the Internet Engineering Task Force (IETF), its areas, and its working groups. Note that other groups may also distribute working documents as Internet- Drafts. Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet- Drafts as reference material or to cite them other than as "work in progress." The list of current Internet-Drafts can be accessed at http://www.ietf.org/ietf/1id-abstracts.txt The list of Internet-Draft Shadow Directories can be accessed at http://www.ietf.org/shadow.html. Abstract 1. Introduction. The goal of this document is to explain the need for congestion control in the Internet, and to discuss what constitutes correct congestion control. One specific goal is to illustrate the dangers of neglecting to apply proper congestion control. A second goal is to discuss the role of the IETF in standardizing new congestion control protocols. This document draws heavily from earlier RFCs, in some cases reproducing entire sections of the text of earlier documents [RFC2309, RFC2357]. We have also borrowed heavily from earlier publications addressing the need for end-to-end congestion control [FF99]. NOTE: This internet-draft is a rough first draft intended Floyd [Page 1]
draft-ieft-floyd-cong-00 October 1999 for further discussion in the IETF (written or gathered-together just before the deadline for submitting internet-drafts to the December IETF). 2. Current standards on congestion control IETF standards concerning end-to-end congestion control focus either on specific protocols (e.g., TCP [RFC2581], reliable multicast protocols [RFC2357]) or on the syntax and semantics of communications between the end nodes and routers about congestion information (e.g., Explicit Congestion Notification [RFC2481]) or desired quality-of- service (diff-serv)). The role of end-to-end congestion control is also discussed in an Informational RFC on ``Recommendations on Queue Management and Congestion Avoidance in the Internet'' [RFC2309]. RFC 2309 recommends the deployment of active queue management mechanisms in routers, and the continuation of design efforts towards mechanisms in routers to deal with flows that are unresponsive to congestion notification. While this note does not address either of the two issues raised by RFC 2309, we freely borrow from RFC 2309 some of their general discussion of end-to-end congestion control. In contrast to the RFCs discussed above, this document is a more general discussion of the principles of congestion control. One of the keys to the success of the Internet has been the congestion avoidance mechanisms of TCP. While TCP is still the dominant transport protocol in the Internet, it is not ubiquitous, and there are an increasing number of applications that, for one reason or another, choose not to use TCP. Such traffic includes not only multicast traffic, but unicast traffic such as streaming multimedia that does not require reliability; and traffic such as DNS or routing messages that consist of short transfers deemed critical to the operation of the network. The continued use of end-to-end congestion control by best-effort traffic is critical for maintaining the stability of the Internet. This document also discusses the general role of the IETF in the standardization of new congestion control protocols. The discussion of congestion control principles for differentiated services or integrated services is a separate issue that is not addressed in this document. Some categories of integrated or differentiated services include a guarantee by the network of end-to- end bandwidth, and as such do not require end-to-end congestion control mechanisms. Floyd [Page 2]
draft-ieft-floyd-cong-00 October 1999 3. The development of end-to-end congestion control. 3.1. Preventing congestion collapse. The Internet protocol architecture is based on a connectionless end- to-end packet service using the IP protocol. The advantages of its connectionless design, flexibility and robustness, have been amply demonstrated. However, these advantages are not without cost: careful design is required to provide good service under heavy load. In fact, lack of attention to the dynamics of packet forwarding can result in severe service degradation or "Internet meltdown". This phenomenon was first observed during the early growth phase of the Internet of the mid 1980s [RFC896], and is technically called "congestion collapse". The original specification of TCP [RFC793] included window-based flow control as a means for the receiver to govern the amount of data sent by the sender. This flow control was used to prevent overflow of the receiver's data buffer space available for that connection. [RFC793] reported that segments could be lost due either to errors or to network congestion, but did not include dynamic adjustment of the flow-control window in response to congestion. The original fix for Internet meltdown was provided by Van Jacobson. Beginning in 1986, Jacobson developed the congestion avoidance mechanisms that are now required in TCP implementations [Jacobson88, RFC 2581]. These mechanisms operate in the hosts to cause TCP connections to "back off" during congestion. We say that TCP flows are "responsive" to congestion signals (i.e., dropped packets) from the network. It is these TCP congestion avoidance algorithms that prevent the congestion collapse of today's Internet. However, that is not the end of the story. Considerable research has been done on Internet dynamics since 1988, and the Internet has grown. It has become clear that the TCP congestion avoidance mechanisms [RFC2581], while necessary and powerful, are not sufficient to provide good service in all circumstances. In addition to the development of new congestion control mechanisms [RFC2357], router-based mechanisms are in development that complement the endpoint congestion avoidance mechanisms. A major issue that still needs to be addressed is the potential for future congestion collapse of the Internet due to flows that do not use responsible end-to-end congestion control. Floyd [Page 3]
draft-ieft-floyd-cong-00 October 1999 3.2. Fairness In addition to a concern about congestion collapse, there is a concern about ``fairness'' for best-effort traffic. Because TCP "backs off" during congestion, a large number of TCP connections can share a single, congested link in such a way that bandwidth is shared reasonably equitably among similarly situated flows. The equitable sharing of bandwidth among flows depends on the fact that all flows are running basically the same congestion avoidance algorithms, conformant with the current TCP specification [RFC793, RFC1122, RFC2581]. The issue of fairness among competing TCP flows has become increasingly important for several reasons. First, using window scaling [RFC1323], individual TCPs can use high bandwidth even over high-propagation-delay paths. Second, with the growth of the web, , Internet users increasing want high-bandwidth and low-delay communications (as opposed to the leisurely transfer of a long file in the background). The growth of best-effort traffic that does not use TCP underscores this concern about fairness between competing best-effort traffic in times of congestion. The popularity of the Internet has caused a proliferation in the number of TCP implementations. Some of these may fail to implement the TCP congestion avoidance mechanisms correctly because of poor implementation [RFC2525]. Others may deliberately be implemented with congestion avoidance algorithms that are more aggressive in their use of bandwidth than other TCP implementations; this would allow a vendor to claim to have a "faster TCP". The logical consequence of such implementations would be a spiral of increasingly aggressive TCP implementations, leading back to the point where there is effectively no congestion avoidance and the Internet is chronically congested. There is a well-known way to achieve more aggressive TCP performance without even changing TCP, by changing the level of granularity: open multiple connections to the same place, as has been done in some Web browsers. Thus, instead of a spiral of increasingly aggressive transport protocols, we could instead have a spiral of increasingly aggressive web browsers, or increasingly aggressive transport protocols. This raises the issue of the appropriate granularity of a "flow", where we define a ``flow'' as the level of granularity appropriate for the application of both fairness and congestion control.. From RFC 2309: ``There are a few "natural" answers: 1) a TCP or UDP connection (source address/port, destination address/port); 2) a source/destination host pair; 3) a given source host or a given Floyd [Page 4]
draft-ieft-floyd-cong-00 October 1999 destination host. We would guess that the source/destination host pair gives the most appropriate granularity in many circumstances. The granularity of flows for congestion management is, at least in part, a policy question that needs to be addressed in the wider IETF community.'' Again borrowing from RFC 2309, we use the term "TCP-compatible" for a flow that behaves under congestion like a flow produced by a conformant TCP. A TCP-compatible flow is responsive to congestion notification, and in steady-state it uses no more bandwidth than a conformant TCP running under comparable conditions (drop rate, RTT, MTU, etc.) It is convenient to divide flows into three classes: (1) TCP- compatible flows, (2) unresponsive flows, i.e., flows that do not slow down when congestion occurs, and (3) flows that are responsive but are not TCP-compatible. The last two classes contain more aggressive flows that pose significant threats to Internet performance, as we discuss below. 3.3. Optimizing performance regarding throughput, delay, and loss. In addition to the prevention of congestion collapse and concerns about fairness, a third reason for a flow to use end-to-end congestion control can be to optimize its own performance regarding throughput, delay, and loss. In some circumstances, for example in environments of high statistical multiplexing, the delay and loss rate experienced by a flow might be largely independent of its own sending rate. However, in environments with lower levels of statistical multiplexing or with per-flow scheduling, the delay and loss rate experienced by a flow can be in part a function of the flow's own sending rate. Thus, a flow can use end-to-end congestion control to limit the delay or loss experienced by its own packets. We would note, however, that in an environment like the current best- effort Internet, concerns regarding congestion collapse and fairness with competing flows limit the range of congestion control behaviors available to a flow. 4. The role of the standards process The standardization of a transport protocol includes not only standardization of aspects of the protocol that could affect interoperability (e.g., information exchanged by the end-nodes), but also standardization of mechanisms deemed critical to performance (e.g., in TCP, reduction of the congestion window in response to a packet drop). At the same time, implementation-specific details and other aspects of the transport protocol that do not affect interoperability and do not significantly interfere with performance Floyd [Page 5]
draft-ieft-floyd-cong-00 October 1999 do not require standardization. Areas of TCP that do not require standardization include the details of TCP's Fast Recovery procedure after a Fast Retransmit [RFC2582]. The appendix uses examples from TCP to discuss in more detail the role of the standards process in the development of congestion control. In addition to addressing the danger of congestion collapse, the standardization process for new transport protocols takes care to avoid a congestion control ``arms race'' among competing protocols. As an example, in RFC 2357 [RFC2357] the TSV Area Directors and their Directorate outline criteria for the publication as RFCs of Internet- Drafts on reliable multicast transport protocols. From [RFC2357]: ``A particular concern for the IETF is the impact of reliable multicast traffic on other traffic in the Internet in times of congestion, in particular the effect of reliable multicast traffic on competing TCP traffic.... The challenge to the IETF is to encourage research and implementations of reliable multicast, and to enable the needs of applications for reliable multicast to be met as expeditiously as possible, while at the same time protecting the Internet from the congestion disaster or collapse that could result from the widespread use of applications with inappropriate reliable multicast mechanisms.'' The list of technical criteria that must be addressed by RFCs on new reliable multicast transport protocols include the following: ``Is there a congestion control mechanism? How well does it perform? When does it fail? Note that congestion control mechanisms that operate on the network more aggressively than TCP will face a great burden of proof that they don't threaten network stability.'' It is reasonable to expect that these concerns about the effect of new transport protocols on competing traffic will apply not only to reliable multicast protocols, but to unreliable unicast, reliable unicast, and unreliable multicast traffic as well. 5. A description of congestion collapse This section discusses congestion collapse from undelivered packets in some detail, and shows how unresponsive flows could contribute to congestion collapse in the Internet. This section draws heavily on material from [FF99]. Informally, congestion collapse occurs when an increase in the network load results in a decrease in the useful work done by the network. As discussed in Section 3, congestion collapse was first reported in the mid 1980s [RFC896], and was largely due to TCP connections unnecessarily retransmitting packets that were either in transit or had already been received at the receiver. We call the Floyd [Page 6]
draft-ieft-floyd-cong-00 October 1999 congestion collapse that results from the unnecessary retransmission of packets classical congestion collapse. Classical congestion collapse is a stable condition that can result in throughput that is a small fraction of normal [RFC896]. Problems with classical congestion collapse have generally been corrected by the timer improvements and congestion control mechanisms in modern implementations of TCP [Jacobson88]. A second form of potential congestion collapse occurs due to undelivered packets. Congestion collapse from undelivered packets arises when bandwidth is wasted by delivering packets through the network that are dropped before reaching their ultimate destination. This is probably the largest unresolved danger with respect to congestion collapse in the Internet today. Different scenarios can result in different degrees of congestion collapse, in terms of the fraction of the congested links' bandwidth used for productive work. The danger of congestion collapse from undelivered packets is due primarily to the increasing deployment of open-loop applications not using end-to-end congestion control. Even more destructive would be best-effort applications that *increase* their sending rate in response to an increased packet drop rate (e.g., automatically using an increased level of FEC). Table 1 gives the results from a scenario with congestion collapse from undelivered packets, where scarce bandwidth is wasted by packets that never reach their destination. The simulation uses a scenario with three TCP flows and one UDP flow competing over a congested 1.5 Mbps link. The access links for all nodes are 10 Mbps, except that the access link to the receiver of the UDP flow is 128 Kbps, only 9 of the bandwidth of shared link. When the UDP source rate exceeds 128 Kbps, most of the UDP packets will be dropped at the output port to that final link. Floyd [Page 7]
draft-ieft-floyd-cong-00 October 1999 UDP Arrival UDP TCP Total Rate Goodput Goodput Goodput -------------------------------------- 0.7 0.7 98.5 99.2 1.8 1.7 97.3 99.1 2.6 2.6 96.0 98.6 5.3 5.2 92.7 97.9 8.8 8.4 87.1 95.5 10.5 8.4 84.8 93.2 13.1 8.4 81.4 89.8 17.5 8.4 77.3 85.7 26.3 8.4 64.5 72.8 52.6 8.4 38.1 46.4 58.4 8.4 32.8 41.2 65.7 8.4 28.5 36.8 75.1 8.4 19.7 28.1 87.6 8.4 11.3 19.7 105.2 8.4 3.4 11.8 131.5 8.4 2.4 10.7 Table 1. A simulation with three TCP flows and one UDP flow. Table 1 shows the UDP arrival rate from the sender, the UDP goodput (defined as the bandwidth delivered to the receiver), the TCP goodput (as delivered to the TCP receivers), and the aggregate goodput on the congested 1.5 Mbps link. Each rate is given as a fraction of the bandwidth of the congested link. As the UDP source rate increases, the TCP goodput decreases roughly linearly, and the UDP goodput is nearly constant. Thus, as the UDP flow increases its offered load, its only effect is to hurt the TCP and aggregate goodput. On the congested link, the UDP flow ultimately ``wastes'' the bandwidth that could have been used by the TCP flow, and reduces the goodput in the network as a whole down to a small fraction of the bandwidth of the congested link. The simulations in Table 1 illustrate both unfairness and congestion collapse. As [FF99] discusses, compatible congestion control is not the only way to provide fairness; per-flow scheduling at the congested routers is an alternative mechanism at the routers that guarantees fairness. However, as discussed in [FF99], per-flow scheduling can not be relied upon to prevent congestion collapse. There are only two alternatives for eliminating the danger of congestion collapse from undelivered packets. The first alternative for preventing congestion collapse from undelivered packets is the use of effective end-to-end congestion control by the end nodes. More specifically, the requirement would be that a flow avoid a pattern of significant losses at links downstream from the first Floyd [Page 8]
draft-ieft-floyd-cong-00 October 1999 congested link on the path. (Here, we would consider any link a ``congested link'' if any flow is using bandwidth that would otherwise be used by other traffic on the link.) Given that an end- node is generally unable to distinguish between a path with one congested link and a path with multiple congested links, the most reliable way for a flow to avoid a pattern of significant losses at a downstream congested link is for the flow to use end-to-end congestion control, and reduce its sending rate in the presence of loss. A second alternative for preventing congestion collapse from undelivered packets would be a guarantee by the network that packets accepted at a congested link in the network will be delivered all the way to the receiver. We note that the choice between the first alternative of end-to-end congestion control and the second alternative of end-to-end bandwidth guarantees does not have to be an either/or decision; congestion collapse can be prevented by the use of effective end-to-end congestion by some of the traffic, and the use of end-to-end bandwidth guarantees from the network for the rest of the traffic. 6. Forms of end-to-end congestion control This document has discussed extensively concerns about congestion collapse and about fairness of new forms of congestion control with TCP. This does not mean, however, that concerns about congestion collapse and fairness with TCP necessitate that all best-effort traffic deploy congestion control based on TCP's Additive-Increase Multiplicative-Decrease (AIMD) algorithm of reducing the sending rate in half in response to each packet drop. This section separately discusses the implications of these two concerns of congestion collapse and fairness with TCP. 6.1. End-to-end congestion control for avoiding congestion collapse. The avoidance of congestion collapse from undelivered packets requires that flows avoid a scenario of a high sending rate, multiple congested links, and a persistent high packet drop rate at the downstream link. Because congestion collapse from undelivered packets consists of packets that waste valuable bandwidth only to be dropped downstream, this form of congestion collapse is not possible in an environment where each flow traverses only one congested link, or where only a small number of packets are dropped at links downstream of the first congested link. Thus, any form of congestion control that successfully avoids a high sending rate in the presence of a high packet drop rate should be sufficient to avoid congestion collapse from undelivered packets. Floyd [Page 9]
draft-ieft-floyd-cong-00 October 1999 We would note that the addition of Explicit Congestion Notification (ECN) to the IP architecture would not, in and of itself, remove the danger of congestion collapse for best-effort traffic. ECN is an experimental proposal that would allow routers to set a bit in packet headers as an indication of congestion to the end-nodes, rather than being forced to rely on packet drops to indicate congestion. However, with ECN, packet-marking would replace packet-dropping only in times of moderate congestion. In particular, when congestion is heavy, and a router's buffers overflow, the router has no choice but to drop arriving packets. 6.2. End-to-end congestion control for fairness with TCP. The concern expressed in [RFC2357] about fairness with TCP places a significant though not crippling constraint on the range of viable end-to-end congestion control mechanisms for best-effort traffic. An environment with per-flow scheduling at all congested links would isolate flows from each other, and eliminate the need for congestion control mechanisms to be TCP-compatible. An environment with differentiated services and some form of class-based scheduling, where flows marked as belonging to a certain diff-serv class would be scheduled in isolation from best-effort traffic, could allow the emergence of an entire diff-serv class of traffic where congestion control was not required to be TCP-compatible. Similarly, a pricing- controlled environment, or a diff-serv class with its own pricing paradigm, could supercede the concern about fairness with TCP. However, for the current Internet environment, where other best- effort traffic could compete in a FIFO queue with TCP traffic, the absence of fairness with TCP could lead to one flow ``starving out'' another flow in a time of high congestion, as was illustrated in Table 1 above. However, the list of TCP-compatible congestion control procedures is not limited to AIMD with the same increase/ decrease parameters as TCP. Other TCP-compatible congestion control procedures include rate-based variants of AIMD; AIMD with different sets of increase/decrease parameters; equation-based congestion control where the sender adjusts its sending rate in response to information about the long-term packet drop rate; layered multicast where receivers subscribe and unsubscribe from layered multicast groups; and possibly other forms that we have not yet begun to consider. Floyd [Page 10]
draft-ieft-floyd-cong-00 October 1999 7. Acknowledgements Much of this document draws directly on previous RFCs addressing end- to-end congestion control. This attempts to be a summary of ideas that have been discussed for many years, and by many people. In particular, acknowledgement is due to the members of the End-to-End Research Group, the Reliable Multicast Research Group, and the Transport Area Directorate. Floyd [Page 11]
draft-ieft-floyd-cong-00 October 1999 8. References [FF98] Floyd, S., and Fall, K., Promoting the Use of End-to-End Congestion Control in the Internet. IEEE/ACM Transactions on Networking, August 1999. URL http://www- nrg.ee.lbl.gov/floyd/end2end-paper.html". [Jacobson88] V. Jacobson, Congestion Avoidance and Control, ACM SIGCOMM '88, August 1988. [RFC793] J. Postel, Transmission Control Protocol," RFC 793, September 1981. [RFC896] Nagle, J., Congestion Control in IP/TCP", RFC 896, January 1984. [RFC1122] Braden, R., Ed., Requirements for Internet Hosts -- Communication Layers, STD 3, RFC 1122, October 1989. [RFC2119] S. Bradner, Key words for use in RFCs to Indicate Requirement Levels, RFC 2119, March 1997. [RFC2309] B. Braden, Clark, D., Crowcroft, J., Davie, B., S. Deering, Estrin, D., Floyd, S., V. Jacobson, G. Minshall, Partridge, C., L. Peterson, Ramakrishnan, K.K., Shenker, S., Wroclawski, J., and Zhang, L., Recommendations on Queue Management and Congestion Avoidance in the Internet, RFC 2309, April 1998. [RFC2357] A. Mankin, A. Romanow, S. Bradner, and V. Paxson, IETF Criteria for Evaluating Reliable Multicast Transport and Application Protocols, RFC 2357, June 1998. [RFC2414] Allman, M., Floyd, S., and Partridge, C., Increasing TCP's Initial Window, RFC 2414, Experimental, September 1998. [RFC2481] K. Ramakrishnan and S. Floyd, A Proposal to add Explicit Congestion Notification (ECN) to IP, RFC 2481, January 1999. [RFC2525] V. Paxson, M Allman, S. Dawson, W. Fenner, and J. Griner, I. Heavens, K. Lahey, J. Semke, B. Volz, Known TCP Implementation Problems, RFC 2525, March 1999. [RFC2581] M. Allman, V. Paxson, and W. Stevens, TCP Congestion Control, RFC 2581, April 1999. [RFC2582] Floyd, S., and Henderson, T., The NewReno Modification to TCP's Fast Recovery Algorithm, RFC 2582, April 1999. Floyd [Page 12]
draft-ieft-floyd-cong-00 October 1999 9. TCP-Specific issues In this section we discuss some of the particulars of TCP congestion control, to illustrate a realization of the congestion control principles, including some of the details that arise when incorporating them into a production transport protocol. [We re-emphasize that is a rough draft of a document to be discussed in the IETF. This is not necessarily a reflection of a rough consensus in the IETF.] 9.1. Slow-start. The TCP sender can not open a new connection by sending a large burst of data (e.g., a receiver's advertised window) all at once. The TCP sender is limited by a small initial value for the congestion window. During slow-start, the TCP sender can increase its sending rate by at most a factor of two in one roundtrip time. Slow-start ends when the sender's congestion window is greater than the slow-start threshold ssthresh. An issue that potentially affects global congestion control, and therefore has been explicitly addressed in the standards process, includes an increase in the value of the initial window [RFC2414,RFC2581]. Issues that have not been addressed in the standards process, and are generally considered not to require standardization, include such issues as the use (or non-use) of rate-based pacing, and mechanisms for ending slow-start early, before the congestion window reaches ssthresh. 9.2. Additive Increase, Multiplicative Decrease. In the absence of congestion, the TCP sender increases its congestion window by at most one packet per roundtrip time (or more precisely, by at most 1/W packets for each ACK received). In response to a packet drop, the TCP sender decreases its congestion window by half. (More precisely, the new congestion window is half of the minimum of the congestion window and the receiver's advertised window.) An issue that potentially affects global congestion control, and therefore would be likely to be explicitly addressed in the standards process, would include a proposed addition of congestion control for the return stream of ``pure acks''. A similar issue would be a proposal for increasing TCP's congestion window based on the number of bytes acked, rather than on the number of acknowledgements received; Floyd [Page 13]
draft-ieft-floyd-cong-00 October 1999 An issue that has not been addressed in the standards process, and is generally not considered to require standardization, would be a change to the congestion window to apply as an upper bound on the number of bytes presumed to be in the pipe, instead of applying as a sliding window starting from the cumulative acknowledgement. (Clearly, the receiver's advertised window applies as a sliding window starting from the cumulative acknowledgement field, because packets received above the cumulative acknowledgement field are held in TCP's receive buffer, and have not been delivered to the application. However, the congestion window applies to the number of packets outstanding in the pipe, and does not necessarily have to include packets that have been received out-of-order by the TCP receiver.) 9.3. Retransmit timers. The TCP sender sets a retransmit timer to infer that a packet has been dropped in the network. When the retransmit timer expires, the sender infers that a packet has been lost, sets ssthresh to half of the current window, and goes into slow-start, retransmitting the lost packet. If the retransmit timer expires because no acknowledgement has been received for a retransmitted packet, the retransmit timer is also "backed-off", doubling the value of the next retransmit timeout interval. An issue that potentially affects global congestion control, and therefore would be likely to be explicitly addressed in the standards process, might include a modified mechanism for setting the retransmit timer that could significantly increase the number of retransmit timers that expire prematurely, when the acknowledgement has not yet arrived at the sender, but in fact no packets have been dropped. This could be of concern to the Internet standards process because retransmit timers that expire prematurely could lead to an increase in the number of packets unnecessarily transmitted on a congested link. An issue that has not been addressed in the standards process, and would be less likely to require standardization, would be a proposed change to algorithms for setting the retransmit timer that would not be expected to significantly increase the chance of the retransmit timer expiring prematurely. 9.4. Fast Retransmit and Fast Recovery. After seeing three duplicate acknowledgements, the TCP sender infers a packet loss. The TCP sender sets ssthresh to half of the current window, reduces the congestion window to at most half of the previous window, and retransmits the lost packet. Floyd [Page 14]
draft-ieft-floyd-cong-00 October 1999 An issue that potentially affects global congestion control, and therefore would be likely to be explicitly addressed in the standards process, might include a proposal (if there was one) for inferring a lost packet after only one or two duplicate acknowledgements. If poorly designed, such a proposal could lead to an increase in the number of packets unnecessarily transmitted on a congested path. An issue that has not been addressed in the standards process, and would not be expected to require standardization, would be a proposal to send a "new" or presumed-lost packet in response to a duplicate or partial acknowledgement, if allowed by the congestion window. An example of this would be sending a new packet in response to a single duplicate acknowledgement, to keep the ``ack clock'' going in case no further acknowledgements would have arrived. 9.5. Other aspects of TCP congestion control. Other aspects of TCP congestion control that have not been discussed in any of the sections above include TCP's recovery from an idle or application-limited period, as well as the entire set of issues raised by the proposal for Explicit Congestion Management. 10. Security Considerations AUTHORS' ADDRESSES Sally Floyd AT&T Center for Internet Research at ICSI (ACIRI) Phone: +1 (510) 642-4274 x189 Email: floyd@aciri.org URL: http://www.aciri.org/floyd/ This draft was created in October 1999. It expires April 2000. Floyd [Page 15]
