Internet DRAFT - draft-ietf-tsvwg-initwin
draft-ietf-tsvwg-initwin
Internet Engineering Task Force Mark Allman
INTERNET DRAFT BBN/NASA GRC
File: draft-ietf-tsvwg-initwin-04.txt Sally Floyd
ICIR
Craig Partridge
BBN Technologies
June, 2002
Expires: December, 2002
Increasing TCP's Initial Window
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
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The list of current Internet-Drafts can be accessed at
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The list of Internet-Draft Shadow Directories can be accessed at
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Abstract
This document specifies an optional standard for TCP to increase the
permitted initial window from one or two segment(s) to roughly 4K bytes,
replacing RFC 2414. This document discusses the advantages and
disadvantages of the higher initial window. The document includes
discussion of experiments and simulations showing that the higher
initial window does not lead to congestion collapse. Finally, the
document provides guidance on implementation issues.
Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC 2119 [RFC2119].
1. TCP Modification
This document updates [RFC2414] and specifies an increase in the
permitted upper bound for TCP's initial window from one or two
segment(s) to between two and four segments. In most cases, this
change results in an upper bound on the initial window of roughly 4K
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bytes (although given a large segment size, the permitted initial
window of two segments may be significantly larger than 4K bytes).
The upper bound for the initial window is given more precisely in
(1):
min (4*MSS, max (2*MSS, 4380 bytes)) (1)
Note: Sending a 1500 byte packet indicates an MSS of 1460 bytes
(assuming no IP or TCP options). Therefore, limiting the initial
window's MSS to 4380 bytes allows the sender to transmit three
segments initially in the common case when using 1500 byte packets.
Equivalently, the upper bound for the initial window size is based
on the maximum segment size (MSS), as follows:
If (MSS <= 1095 bytes)
then win <= 4 * MSS;
If (1095 bytes < MSS < 2190 bytes)
then win <= 4380;
If (2190 bytes <= MSS)
then win <= 2 * MSS;
This increased initial window is optional: a TCP MAY start with a
larger initial window. However, we expect that most general-purpose
TCP implementations would choose to use the larger initial
congestion window given in equation (1) above.
This upper bound for the initial window size represents a change
from RFC 2581 [RFC2581], which specified that the congestion window
be initialized to one or two segments.
This change applies to the initial window of the connection in the
first round trip time (RTT) of data transmission following the TCP
three- way handshake. Neither the SYN/ACK nor its acknowledgment
(ACK) in the three-way handshake should increase the initial window
size above that outlined in equation (1). If the SYN or SYN/ACK is
lost, the initial window used by a sender after a correctly
transmitted SYN MUST be one segment consisting of MSS bytes.
TCP implementations use slow start in as many as three different
ways: (1) to start a new connection (the initial window); (2) to
restart transmission after a long idle period (the restart window);
and (3) to restart transmission after a retransmit timeout (the loss
window). The change specified in this document affects the value of
the initial window. Optionally, a TCP MAY set the restart window to
the minimum of the value used for the initial window and the current
value of cwnd (in other words, using a larger value for the restart
window should never increase the size of cwnd). These changes do
NOT change the loss window, which must remain 1 segment of MSS bytes
(to permit the lowest possible window size in the case of severe
congestion).
2. Implementation Issues
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When larger initial windows are implemented along with Path MTU
Discovery [RFC1191], and the MSS being used is found to be too
large, the congestion window `cwnd' SHOULD be reduced to prevent
large bursts of smaller segments. Specifically, `cwnd' SHOULD be
reduced by the ratio of the old segment size to the new segment
size.
When larger initial windows are implemented along with Path MTU
Discovery [RFC1191], alternatives are to set the "Don't Fragment"
(DF) bit in all segments in the initial window, or to set the "Don't
Fragment" (DF) bit in one of the segments. It is an open question
which of these two alternatives is best; we would hope that
implementation experiences will shed light on this question. In the
first case of setting the DF bit in all segments, if the initial
packets are too large, then all of the initial packets will be
dropped in the network. In the second case of setting the DF bit in
only one segment, if the initial packets are too large, then all but
one of the initial packets will be fragmented in the network. When
the second case is followed, setting the DF bit in the last segment
in the initial window provides the least chance for needless
retransmissions when the initial segment size is found to be too
large, because it minimizes the chances of duplicate ACKs triggering
a Fast Retransmit. However, more attention needs to be paid to the
interaction between larger initial windows and Path MTU Discovery.
The larger initial window specified in this document is not intended
as encouragement for web browsers to open multiple simultaneous TCP
connections all with large initial windows. When web browsers open
simultaneous TCP connections to the same destination, this works
against TCP's congestion control mechanisms [FF98], regardless of
the size of the initial window. Combining this behavior with larger
initial windows further increases the unfairness to other traffic in
the network. We suggest the use of HTTP/1.1 [RFC2068] (persistent
TCP connections and pipelining) as a way to achieve better
performance of web transfers.
3. Advantages of Larger Initial Windows
1. When the initial window is one segment, a receiver employing
delayed ACKs [RFC1122] is forced to wait for a timeout before
generating an ACK. With an initial window of at least two
segments, the receiver will generate an ACK after the second
data segment arrives. This eliminates the wait on the timeout
(often up to 200 msec, and possibly up to 500 msec [RFC1122]).
2. For connections transmitting only a small amount of data, a
larger initial window reduces the transmission time (assuming at
most moderate segment drop rates). For many email (SMTP
[Pos82]) and web page (HTTP [RFC1945, RFC2068]) transfers that
are less than 4K bytes, the larger initial window would reduce
the data transfer time to a single RTT.
3. For connections that will be able to use large congestion
windows, this modification eliminates up to three RTTs and a
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delayed ACK timeout during the initial slow-start phase. This
will be of particular benefit for high-bandwidth large-
propagation-delay TCP connections, such as those over satellite
links.
4. Disadvantages of Larger Initial Windows for the Individual
Connection
In high-congestion environments, particularly for routers that have
a bias against bursty traffic (as in the typical Drop Tail router
queues), a TCP connection can sometimes be better off starting with
an initial window of one segment. There are scenarios where a TCP
connection slow-starting from an initial window of one segment might
not have segments dropped, while a TCP connection starting with an
initial window of four segments might experience unnecessary
retransmits due to the inability of the router to handle small
bursts. This could result in an unnecessary retransmit timeout.
For a large-window connection that is able to recover without a
retransmit timeout, this could result in an unnecessarily-early
transition from the slow-start to the congestion-avoidance phase of
the window increase algorithm. These premature segment drops are
unlikely to occur in uncongested networks with sufficient buffering
or in moderately-congested networks where the congested router uses
active queue management (such as Random Early Detection [FJ93,
RFC2309]).
Some TCP connections will receive better performance with the larger
initial window even if the burstiness of the initial window results
in premature segment drops. This will be true if (1) the TCP
connection recovers from the segment drop without a retransmit
timeout, and (2) the TCP connection is ultimately limited to a small
congestion window by either network congestion or by the receiver's
advertised window.
5. Disadvantages of Larger Initial Windows for the Network
In terms of the potential for congestion collapse, we consider two
separate potential dangers for the network. The first danger would
be a scenario where a large number of segments on congested links
were duplicate segments that had already been received at the
receiver. The second danger would be a scenario where a large
number of segments on congested links were segments that would be
dropped later in the network before reaching their final
destination.
In terms of the negative effect on other traffic in the network, a
potential disadvantage of larger initial windows would be that they
increase the general packet drop rate in the network. We discuss
these three issues below.
Duplicate segments:
As described in the previous section, the larger initial window
could occasionally result in a segment dropped from the initial
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window, when that segment might not have been dropped if the
sender had slow-started from an initial window of one segment.
However, Appendix A shows that even in this case, the larger
initial window would not result in the transmission of a large
number of duplicate segments.
Segments dropped later in the network:
How much would the larger initial window for TCP increase the
number of segments on congested links that would be dropped
before reaching their final destination? This is a problem that
can only occur for connections with multiple congested links,
where some segments might use scarce bandwidth on the first
congested link along the path, only to be dropped later along
the path.
First, many of the TCP connections will have only one congested
link along the path. Segments dropped from these connections do
not "waste" scarce bandwidth, and do not contribute to
congestion collapse.
However, some network paths will have multiple congested links,
and segments dropped from the initial window could use scarce
bandwidth along the earlier congested links before ultimately
being dropped on subsequent congested links. To the extent that
the drop rate is independent of the initial window used by TCP
segments, the problem of congested links carrying segments that
will be dropped before reaching their destination will be
similar for TCP connections that start by sending four segments
or one segment.
An increased packet drop rate:
For a network with a high segment drop rate, increasing the TCP
initial window could increase the segment drop rate even
further. This is in part because routers with Drop Tail queue
management have difficulties with bursty traffic in times of
congestion. However, given uncorrelated arrivals for TCP
connections, the larger TCP initial window should not
significantly increase the segment drop rate. Simulation-based
explorations of these issues are discussed in Section 7.2.
These potential dangers for the network are explored in simulations
and experiments described in the section below. Our judgment is
that while there are dangers of congestion collapse in the current
Internet (see [FF98] for a discussion of the dangers of congestion
collapse from an increased deployment of UDP connections without
end-to-end congestion control), there is no such danger to the
network from increasing the TCP initial window to 4K bytes.
6. Interactions with the Retransmission Timer
Using a larger initial burst of data can exacerbate existing
problems with spurious retransmit timeouts on low-bandwidth paths,
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assuming the standard algorithm for determining the TCP
retransmission timeout (RTO) [RFC2988]. The problem is that across
low-bandwidth network paths on which the transmission time of a
packet is a large portion of the round-trip time, the small packets
used to establish a TCP connection do not seed the RTO estimator
appropriately. When the first window of data packets is
transmitted, the sender's retransmit timer could expire before the
acknowledgments for those packets are received. As each
acknowledgment arrives, the retransmit timer is generally reset.
Thus, the retransmit timer will not expire as long as an
acknowledgment arrives at least once a second, given the one-second
minimum on the RTO recommended in RFC 2988.
For instance, consider a 9.6 Kbps link. The initial RTT measurement
will be on the order of 67 msec, if we simply consider the
transmission time of 2 packets (the SYN and SYN-ACK) each consisting
of 40 bytes. Using the RTO estimator given in [RFC2988], this
yields an initial RTO of 201 msec (67 + 4*(67/2)). However, we
round the RTO to 1 second as specified in RFC 2988. Then assume we
send an initial window of one or more 1500-byte packets (1460 data
bytes plus overhead). Each packet will take on the order of 1.25
seconds to transmit. Therefore, the RTO will fire before the ACK
for the first packet returns, causing a spurious timeout. In this
case, a larger initial window of three or four packets exacerbates
the problems caused by this spurious timeout.
One way to deal with this problem is to make the RTO algorithm more
conservative. During the initial window of data, for instance, the
RTO could be updated for each acknowledgment received. In addition,
if the retransmit timer expires for some packet lost in the first
window of data, we could leave the exponential-backoff of the
retransmit timer engaged until at least one valid RTT measurement is
received that involves a data packet.
Another method would be to refrain from taking a RTT sample during
connection establishment, leaving the default RTO in place until TCP
takes a sample from a data segment and the corresponding ACK. While
this method likely helps prevent spurious retransmits it also may
slow the data transfer down if loss occurs before the RTO is
seeded. The use of limited transmit [RFC3042] to aid a TCP
connection in recovering from loss using fast retransmit rather than
the RTO timer mitigates the performance degradation caused by using
the high default RTO during the initial window of data
transmission.
This specification leaves the decision about what to do (if
anything) with regards to the RTO when using a larger initial window
to the implementer. However, the RECOMMENDED approach is to refrain
from sampling the RTT during the three-way handshake, keeping the
default RTO in place until a RTT sample involving a data packet is
taken. In addition, it is RECOMMENDED that TCPs use limited
transmit [RFC3042].
7. Typical Levels of Burstiness for TCP Traffic.
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Larger TCP initial windows would not dramatically increase the
burstiness of TCP traffic in the Internet today, because such
traffic is already fairly bursty. Bursts of two and three segments
are already typical of TCP [Flo97]; A delayed ACK (covering two
previously unacknowledged segments) received during congestion
avoidance causes the congestion window to slide and two segments to
be sent. The same delayed ACK received during slow start causes the
window to slide by two segments and then be incremented by one
segment, resulting in a three-segment burst. While not necessarily
typical, bursts of four and five segments for TCP are not rare.
Assuming delayed ACKs, a single dropped ACK causes the subsequent
ACK to cover four previously unacknowledged segments. During
congestion avoidance this leads to a four-segment burst and during
slow start a five-segment burst is generated.
There are also changes in progress that reduce the performance
problems posed by moderate traffic bursts. One such change is the
deployment of higher-speed links in some parts of the network, where
a burst of 4K bytes can represent a small quantity of data. A
second change, for routers with sufficient buffering, is the
deployment of queue management mechanisms such as RED, which is
designed to be tolerant of transient traffic bursts.
8. Simulations and Experimental Results
8.1 Studies of TCP Connections using that Larger Initial Window
This section surveys simulations and experiments that explore the
effect of larger initial windows on TCP connections. The first set
of experiments explores performance over satellite links. Larger
initial windows have been shown to improve performance of TCP
connections over satellite channels [All97b]. In this study, an
initial window of four segments (512 byte MSS) resulted in
throughput improvements of up to 30% (depending upon transfer size).
[KAGT98] shows that the use of larger initial windows results in a
decrease in transfer time in HTTP tests over the ACTS satellite
system. A study involving simulations of a large number of HTTP
transactions over hybrid fiber coax (HFC) indicates that the use of
larger initial windows decreases the time required to load WWW pages
[Nic97].
A second set of experiments explored TCP performance over dialup
modem links. In experiments over a 28.8 bps dialup channel [All97a,
AHO98], a four-segment initial window decreased the transfer time of
a 16KB file by roughly 10%, with no accompanying increase in the
drop rate. A simulation study [RFC2416] investigated the effects of
using a larger initial window on a host connected by a slow modem
link and a router with a 3 packet buffer. The study concluded that
for the scenario investigated, the use of larger initial windows was
not harmful to TCP performance.
Finally, [All00] illustrates that the percentage of connections at a
particular web server that experience loss in the initial window of
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data transmission increases with the size of the initial congestion
window. However, the increase is in line with what would be
expected from sending a larger burst into the network.
8.2 Studies of Networks using Larger Initial Windows
This section surveys simulations and experiments investigating the
impact of the larger window on other TCP connections sharing the
path. Experiments in [All97a, AHO98] show that for 16 KB transfers
to 100 Internet hosts, four-segment initial windows resulted in a
small increase in the drop rate of 0.04 segments/transfer. While
the drop rate increased slightly, the transfer time was reduced by
roughly 25% for transfers using the four-segment (512 byte MSS)
initial window when compared to an initial window of one segment.
A simulation study in [RFC2415] explores the impact of a larger
initial window on competing network traffic. In this investigation,
HTTP and FTP flows share a single congested gateway (where the
number of HTTP and FTP flows varies from one simulation set to
another). For each simulation set, the paper examines aggregate
link utilization and packet drop rates, median web page delay, and
network power for the FTP transfers. The larger initial window
generally resulted in increased throughput, slightly-increased
packet drop rates, and an increase in overall network power. With
the exception of one scenario, the larger initial window resulted in
an increase in the drop rate of less than 1% above the loss rate
experienced when using a one-segment initial window; in this
scenario, the drop rate increased from 3.5% with one-segment initial
windows, to 4.5% with four-segment initial windows. The overall
conclusions were that increasing the TCP initial window to three
packets (or 4380 bytes) helps to improve perceived performance.
Morris [Mor97] investigated larger initial windows in a highly
congested network with transfers of size 20K. The loss rate in
networks where all TCP connections use an initial window of four
segments is shown to be 1-2% greater than in a network where all
connections use an initial window of one segment. This relationship
held in scenarios where the loss rates with one-segment initial
windows ranged from 1% to 11%. In addition, in networks where
connections used an initial window of four segments, TCP connections
spent more time waiting for the retransmit timer (RTO) to expire to
resend a segment than was spent when using an initial window of one
segment. The time spent waiting for the RTO timer to expire
represents idle time when no useful work was being accomplished for
that connection. These results show that in a very congested
environment, where each connection's share of the bottleneck
bandwidth is close to one segment, using a larger initial window can
cause a perceptible increase in both loss rates and retransmit
timeouts.
9. Security Considerations
This document discusses the initial congestion window permitted for
TCP connections. Changing this value does not raise any known new
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security issues with TCP.
10. Conclusion
This document specifies a small change to TCP that will likely be
beneficial to short-lived TCP connections and those over links with
long RTTs (saving several RTTs during the initial slow-start phase).
11. Acknowledgments
We would like to acknowledge Vern Paxson, Tim Shepard, members of
the End-to-End-Interest Mailing List, and members of the IETF TCP
Implementation Working Group for continuing discussions of these
issues for discussions and feedback on this document.
12. References
[AHO98] Mark Allman, Chris Hayes, and Shawn Ostermann, An Evaluation
of TCP with Larger Initial Windows, March 1998. Submitted to
ACM Computer Communication Review. URL:
"http://roland.lerc.nasa.gov/~mallman/papers/initwin.ps".
[All97a] Mark Allman. An Evaluation of TCP with Larger Initial
Windows. 40th IETF Meeting -- TCP Implementations WG.
December, 1997. Washington, DC.
[All97b] Mark Allman. Improving TCP Performance Over Satellite
Channels. Master's thesis, Ohio University, June 1997.
[All00] Mark Allman. A Web Server's View of the Transport Layer. ACM
Computer Communication Review, 30(5), October 2000.
[FF96] Fall, K., and Floyd, S., Simulation-based Comparisons of
Tahoe, Reno, and SACK TCP. Computer Communication Review,
26(3), July 1996.
[FF98] Sally Floyd, Kevin Fall. Promoting the Use of End-to-End
Congestion Control in the Internet. Submitted to IEEE
Transactions on Networking. URL "http://www-
nrg.ee.lbl.gov/floyd/end2end-paper.html".
[FJ93] Floyd, S., and Jacobson, V., Random Early Detection gateways
for Congestion Avoidance. IEEE/ACM Transactions on Networking,
V.1 N.4, August 1993, p. 397-413.
[Flo94] Floyd, S., TCP and Explicit Congestion Notification.
Computer Communication Review, 24(5):10-23, October 1994.
[Flo96] Floyd, S., Issues of TCP with SACK. Technical report,
January 1996. Available from http://www-nrg.ee.lbl.gov/floyd/.
[Flo97] Floyd, S., Increasing TCP's Initial Window. Viewgraphs,
40th IETF Meeting - TCP Implementations WG. December, 1997. URL
"ftp://ftp.ee.lbl.gov/talks/sf-tcp-ietf97.ps".
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[KAGT98] Hans Kruse, Mark Allman, Jim Griner, Diepchi Tran. HTTP
Page Transfer Rates Over Geo-Stationary Satellite Links. March
1998. Proceedings of the Sixth International Conference on
Telecommunication Systems. URL
"http://roland.lerc.nasa.gov/~mallman/papers/nash98.ps".
[Mor97] Robert Morris. Private communication, 1997. Cited for
acknowledgement purposes only.
[Nic97] Kathleen Nichols. Improving Network Simulation with
Feedback. Com21, Inc. Technical Report. Available from
http://www.com21.com/pages/papers/068.pdf.
[Pos82] Postel, J., "Simple Mail Transfer Protocol", STD 10, RFC
821, August 1982.
[RFC1122] Braden, R., "Requirements for Internet Hosts --
Communication Layers", STD 3, RFC 1122, October 1989.
[RFC1191] Mogul, J., and S. Deering, "Path MTU Discovery", RFC 1191,
November 1990.
[RFC1945] Berners-Lee, T., Fielding, R., and H. Nielsen, "Hypertext
Transfer Protocol -- HTTP/1.0", RFC 1945, May 1996.
[RFC2068] Fielding, R., Mogul, J., Gettys, J., Frystyk, H., and T.
Berners-Lee, "Hypertext Transfer Protocol -- HTTP/1.1", RFC
2068, January 1997.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC2309] Braden, B., Clark, D., Crowcroft, J., Davie, B., Deering,
S., Estrin, D., Floyd, S., Jacobson, V., Minshall, G.,
Partridge, C., Peterson, L., Ramakrishnan, K., Shenker, S.,
Wroclawski, J., and L. Zhang, "Recommendations on Queue
Management and Congestion Avoidance in the Internet", RFC 2309,
April 1998.
[RFC2414] Allman, M., Floyd, S., and C. Partridge, "Increasing TCP's
Initial Window", RFC 2414, September 1998.
[RFC2415] Poduri, K., and K. Nichols, "Simulation Studies of
Increased Initial TCP Window Size", RFC 2415, September 1998.
[RFC2416] Shepard, T., and C. Partridge, "When TCP Starts Up With
Four Packets Into Only Three Buffers", RFC 2416, September 1998.
[RFC2581] Mark Allman, Vern Paxson, W. Richard Stevens. TCP
Congestion Control, April 1999. RFC 2581.
[RFC2988] Vern Paxson, Mark Allman. Computing TCP's Retransmission
Timer, November 2000. RFC 2988.
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[RFC3042] M. Allman, H. Balakrishnan, and S. Floyd, Enhancing TCP's
Loss Recovery Using Limited Transmit, RFC 3042, January 2001.
[RFC3168] Ramakrishnan, K.K., Floyd, S., and Black, D., "The
Addition of Explicit Congestion Notification (ECN) to IP", RFC
3168, September 2001.
13. Author's Addresses
Mark Allman
BBN Technologies/NASA Glenn Research Center
21000 Brookpark Road
MS 54-5
Cleveland, OH 44135
EMail: mallman@bbn.com
http://roland.lerc.nasa.gov/~mallman/
Sally Floyd
ICSI Center for Internet Research
1947 Center St, Suite 600
Berkeley, CA 94704
Phone: +1 (510) 666-2989
EMail: floyd@icir.org
http://www.icir.org/floyd/
Craig Partridge
BBN Technologies
10 Moulton Street
Cambridge, MA 02138
EMail: craig@bbn.com
14. Appendix - Duplicate Segments
In the current environment (without Explicit Congestion Notification
[Flo94] [RFC2481]), all TCPs use segment drops as indications from
the network about the limits of available bandwidth. We argue here
that the change to a larger initial window should not result in the
sender retransmitting a large number of duplicate segments that have
already arrived at the receiver.
If one segment is dropped from the initial window, there are three
different ways for TCP to recover: (1) Slow-starting from a window
of one segment, as is done after a retransmit timeout, or after Fast
Retransmit in Tahoe TCP; (2) Fast Recovery without selective
acknowledgments (SACK), as is done after three duplicate ACKs in
Reno TCP; and (3) Fast Recovery with SACK, for TCP where both the
sender and the receiver support the SACK option [MMFR96]. In all
three cases, if a single segment is dropped from the initial window,
no duplicate segments (i.e., segments that have already been
received at the receiver) are transmitted. Note that for a TCP
sending four 512-byte segments in the initial window, a single
segment drop will not require a retransmit timeout, but can be
recovered from using the Fast Retransmit algorithm (unless the
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retransmit timer expires prematurely). In addition, a single
segment dropped from an initial window of three segments might be
repaired using the fast retransmit algorithm, depending on which
segment is dropped and whether or not delayed ACKs are used. For
example, dropping the first segment of a three segment initial
window will always require waiting for a timeout, in the absence of
Limited Transmit [RFC3042]. However, dropping the third segment
will always allow recovery via the fast retransmit algorithm, as
long as no ACKs are lost.
Next we consider scenarios where the initial window contains two to
four segments, and at least two of those segments are dropped. If
all segments in the initial window are dropped, then clearly no
duplicate segments are retransmitted, as the receiver has not yet
received any segments. (It is still a possibility that these
dropped segments used scarce bandwidth on the way to their drop
point; this issue was discussed in Section 5.)
When two segments are dropped from an initial window of three
segments, the sender will only send a duplicate segment if the first
two of the three segments were dropped, and the sender does not
receive a packet with the SACK option acknowledging the third
segment.
When two segments are dropped from an initial window of four
segments, an examination of the six possible scenarios (which we
don't go through here) shows that, depending on the position of the
dropped packets, in the absence of SACK the sender might send one
duplicate segment. There are no scenarios in which the sender sends
two duplicate segments.
When three segments are dropped from an initial window of four
segments, then, in the absence of SACK, it is possible that one
duplicate segment will be sent, depending on the position of the
dropped segments.
The summary is that in the absence of SACK, there are some scenarios
with multiple segment drops from the initial window where one
duplicate segment will be transmitted. There are no scenarios where
more that one duplicate segment will be transmitted. Our conclusion
is that the number of duplicate segments transmitted as a result of
a larger initial window should be small.
15. Full Copyright Statement
Copyright (C) The Internet Society (2001). All Rights Reserved.
This document and translations of it may be copied and furnished to
others, and derivative works that comment on or otherwise explain it
or assist in its implementation may be prepared, copied, published
and distributed, in whole or in part, without restriction of any
kind, provided that the above copyright notice and this paragraph are
included on all such copies and derivative works. However, this
document itself may not be modified in any way, such as by removing
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draft-ietf-tsvwg-initwin-04.txt June 2002
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