One document matched: draft-sridharan-tcpm-ctcp-01.txt
Differences from draft-sridharan-tcpm-ctcp-00.txt
Network Working Group M. Sridharan
Internet Draft Microsoft
Intended status: Experimental K. Tan
October 29, 2007 Microsoft Research
Expires: April 2008 D. Bansal
D. Thaler
Microsoft
Compound TCP: A New TCP Congestion Control for High-Speed and Long
Distance Networks
draft-sridharan-tcpm-ctcp-01.txt
Status of this Memo
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Abstract
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Compound TCP (CTCP) is a modification to TCP's congestion control
mechanism for use with TCP connections with large congestion windows.
This document describes the Compound TCP algorithm in detail, and
solicits experimentation and feedback from the wider community. The
key idea behind CTCP is to add a scalable delay-based component to the
standard TCP's loss-based congestion control. The sending rate of CTCP
is controlled by both loss and delay components. The delay-based
component has a scalable window increasing rule that not only
efficiently uses the link capacity, but on sensing queue build up,
gracefully reduces the sending rate.
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Table of Contents
1. Introduction...................................................3
2. Design Goals...................................................5
3. Compound TCP Control Law.......................................5
4. Compound TCP Response Function.................................8
5. Automatic Selection of Gamma...................................9
6. Implementation Issues.........................................11
7. Deployment Issues.............................................12
8. Security Considerations.......................................13
9. IANA Considerations...........................................13
10. Conclusions..................................................13
11. Acknowledgments..............................................14
12. References...................................................15
12.1. Normative References.......................................15
12.2. Informative References.....................................15
Author's Addresses...............................................16
Intellectual Property Statement..................................17
Disclaimer of Validity...........................................17
1. Introduction
In this document, we collectively refer to any TCP congestion control
algorithm that employs a linear increase function for congestion
control, including TCP Reno and all its variants as Standard TCP. This
document describes Compound TCP, a modification to TCP's congestion
control mechanism for fast, long-distance networks. The standard TCP
congestion avoidance algorithm employs an additive increase and
multiplicative decrease (AIMD) scheme, which employs a conservative
linear growth function for increasing the congestion window and
multiplicative decrease function on encountering a loss. For a high-
speed and long delay network, it takes standard TCP an unreasonably
long time to recover the sending rate after a single loss event
[RFC2581, RFC3649]. Moreover, it is well-known now that in a steady-
state environment, with a packet loss rate of p, the current standard
TCP's average congestion window is inversely proportional to the square
root of the packet loss rate [RFC2581,PADHYE]. Therefore, it requires
an extremely small packet loss rate to sustain a large window. As an
example, Floyd et al. [RFC3649], pointed out that under a 10Gbps link
with 100ms delay, it will roughly take one hour for a standard TCP flow
to fully utilize the link capacity, if no packet is lost or corrupted.
This one hour error-free transmission requires a packet loss rate of
around 10^-11 with 1500-byte size packets (one packet loss over
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2,600,000,000 packet transmission!), which is not practical in today's
networks.
There are several proposals to address this fundamental limitation of
TCP. One straightforward way to overcome this limitation is to modify
TCP's increase/decrease rule in its congestion avoidance stage. More
specifically, in the absence of packet loss, the sender increases
congestion window more quickly and decreases it more gently upon a
packet loss. In a mixed network environment, the aggressive behavior of
such an approache may severely degrade the performance of regular TCP
flows whenever the network path is already highly utilized. When an
aggressive high-speed variant flow traverses the bottleneck link with
other standard TCP flows, it may increase its own share of bandwidth by
reducing the throughput of other competing TCP flows. As a result, the
aggressive variants will cause much more self-induced packet losses on
bottleneck links, and push back the throughput of the regular TCP
flows.
Then there is the class of high-speed protocols which use variances in
RTT as a congestion indicator (e.g., [AFRICA,FAST]). Such delay-based
approaches are more-or-less derived from the seminal work of TCP-Vegas
[VEGAS]. An increase in RTT is considered an early indicator of
congestion, and the sending rate is cut in half to avoid buffer
overflow. The problem in this approach comes when delay-based and loss-
based flows share the same bottleneck link. While the delay-based flows
respond to increases in RTT by cutting its sending rate, the loss-based
flows continue to increase their sending rate. As a result a delay-
based flow obtains far less bandwidth than its fair share. This
weakness is hard to remedy for purely delay-based approaches.
The design of Compound TCP is to satisfy the efficiency requirement and
the TCP friendliness requirement simultaneously. The key idea is that
if the link is under-utilized, the high-speed protocol should be
aggressive and increase the sending rate quickly. However, once the
link is fully utilized, being aggressive will not only adversely affect
standard TCP flows but will also cause instability. As noted above,
delay-based approaches already have the nice property of adjusting
aggressiveness based on the link utilization, which is observed by the
end-systems as an increase in RTT. CTCP incorporates a scalable delay-
based component into the standard TCP's congestion avoidance algorithm.
Using the delay component as an automatic tuning knob, CTCP is scalable
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yet TCP friendly.
2. Design Goals
The design of CTCP is motivated by the following requirements:
o Improve throughput by efficiently using the spare capacity in
the network
o Good intra-protocol fairness when competing with flows that
have different RTTs
o Should not impact the performance of standard TCP flows sharing
the same bottleneck
o No additional feedback or support required from the network
CTCP can efficiently use the network resource and achieve high link
utilization. The aggressiveness can be controlled by adopting a rapid
increase rule in the delay-based component. We choose CTCP to have
similar aggressiveness as HighSpeed TCP [RFC3649]. Our design choice is
motivated by the fact that HSTCP has been tested to be aggressive
enough in real world networks and is now an experimental IETF RFC. We
also wanted an upper bound on the amount of unfairness to standard TCP
flows. However, as shown later, CTCP is able to maintain TCP
friendliness under high statistical multiplexing and also while
traversing poorly buffered links. CTCP has similar or, in some cases,
even improved RTT fairness compared to standard TCP. As we will
demonstrate later this is due to the fact that the amount of backlogged
packets for a connection is independent of the RTT of the connection.
Even though CTCP does not require any feedback from the network, CTCP
works well in ECN capable environments. There is also no expectation on
the queuing algorithm deployed in the routers.
As is the case with most high-speed variants today, CTCP does not
modify slow-start. We agree to the belief that ramping-up faster than
slow-start without additional information from the network can be
harmful. Similar to HSTCP, to ensure TCP compatibility, CTCP's scalable
component uses the same response function as Standard TCP when the
current congestion window is at most Low_Window. CTCP sets Low_Window
to 38 MSS-sized segments, corresponding to a packet drop rate of 10^-3
for TCP.
3. Compound TCP Control Law
CTCP modifies Standard TCP's loss-based control law with a scalable
delay-based component. To do so, a new state variable is introduced in
current TCP Control Block (TCB), namely dwnd (Delay Window), which
controls the delay-based component in CTCP. The conventional congestion
window, cwnd, remains untouched, which controls the loss-based
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component in CTCP. Thus, the CTCP sending window now is controlled by
both cwnd and dwnd. Specifically, the TCP sending window (wnd) is now
calculated as follows:
wnd = min(cwnd + dwnd, awnd), (1)
where awnd is the advertised window from the receiver.
cwnd is updated in the same way as regular TCP in the congestion
avoidance phase, i.e., cwnd is increased by 1 MSS every RTT and halved
when a packet loss is encountered. The update to dwnd will be explained
in detail later in this section. The combined window for CTCP from (1)
above allows up to (cwnd + dwnd) packets in one RTT. Therefore, the
increment of cwnd on the arrival of an ACK is modified accordingly:
cwnd = cwnd + 1/(cwnd+dwnd) (2)
As stated above, CTCP retains the same behavior during slow start. When
a connection starts up, dwnd is initialized to zero while the
connection is in slow start phase. Thus the delay component is
effective when the connection enters congestion avoidance. The delay-
based algorithm has the following properties. It uses a scalable
increase rule when it infers that the network is under-utilized. It
also reduces the sending rate when it senses incipient congestion. By
reducing its sending rate, the delay-based component yields to
competing TCP flows and ensures TCP fairness. It reacts to packet
losses by reducing its sending rate, which is necessary to avoid
congestion collapse. CTCP's control law for the delay-based component
is derived from TCP Vegas. A state variable called basertt tracks the
minimum round trip delay seen by a packet over the network path. When a
connection is started, basertt is updated to be the minimum RTT
observed during the 3-way handshake. basertt is cleared and set to zero
if a retransmission timeout is hit. It is continually measured and
updated to track changing network conditions. The CTCP sender also
maintains a smoothed RTT, srtt, updated as specified in [RFC2988]. The
number of backlogged packets of the connection can then be estimated
using,
expected (throughput) = wnd/basertt
actual (throughput) = wnd/srtt
diff = (expected - actual) * basertt
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The expected throughput gives the estimation of throughput CTCP gets if
it does not overrun (induce queueing on) the network path. The actual
throughput stands for the throughput CTCP really gets. Using this, the
amount of data backlogged in the bottleneck queue (diff) can be
calculated. Congestion is detected by comparing diff to a threshold
gamma. If diff < gamma, the network path is assumed to be under-
utilized; otherwise the network path is assumed to be congested and
CTCP should gracefully reduce its window.
It is to be noted that a connection should have at least gamma packets
backlogged in the bottleneck queue to be able to detect incipient
congestion. This motivates the need for gamma to be small since the
implication is that even when the bottleneck buffer size is small, CTCP
will react early enough to ensure TCP fairness. On the other hand, if
gamma is too small compared to the queue size, CTCP will falsely detect
congestion and will adversely affect the throughput. Choosing the
appropriate value for gamma could be a problem because this parameter
depends on both network configuration and the number of concurrent
flows, which are generally unknown to the end-systems. Section 5
presents an effective way to automatically estimate gamma.
The increase law of the delay-based component should make CTCP more
scalable in high-speed and long delay pipes. We choose a binomial
function to increase the delay window [BAINF01]. As explained in the
next section we have modeled the response function for CTCP to have
comparable scalability to HighSpeed TCP. Since there is already a loss-
based component in CTCP, the delay-based component needs to be designed
to only fill the gap. The control law for CTCP's delay component can be
summarized as follows:
dwnd(t+1) =
dwnd(t) + alpha*dwnd(t)^k - 1, if diff < gamma (3)
dwnd(t) - eta*diff, if diff >= gamma (4)
dwnd(t)(1-beta) - cwnd/2, on packet loss (5)
where alpha, beta, eta and k are tunable to obtain the desirable
scalability, smoothness and responsiveness. Equation (3) shows that in
the increase phase, dwnd only needs to increase by (alpha*dwnd(t)^k -
1) packets, since the loss-based component cwnd will also increase by 1
packet. When a packet loss occurs (detected by three duplicate ACKs),
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dwnd is set to the difference between the desired reduced window size
and that can be provided by cwnd. The rule in equation (4) is very
important to preserve good RTT and TCP fairness. Eta defines how
rapidly the delay component should reduce its window when congestion is
detected. Note that dwnd is never negative, so the CTCP window is lower
bounded by its loss-based component, which is same as Standard TCP.
If a retransmission timeout occurs, dwnd should be reset to zero and
the delay-based component is disabled. This is because after a timeout,
the TCP sender enters slow-start phase. After the CTCP sender exits the
slow-start recovery state and enters congestion avoidance, dwnd control
kicks in again.
4. Compound TCP Response Function
The TCP response function provides a relationship between TCP's average
congestion window w in MSS-sized segments as a function of the steady-
state packet drop rate p. To specify a modified response function for
CTCP, we use the analytical model in [CTCPI06] to derive a relationship
between w and p. Based on this model, the response function for CTCP
provides the following relationship between w and p,
w ~.1/(p^(1/(2-k))) (6)
As explained earlier we modeled the response function for CTCP to have
comparable scalability to HighSpeed TCP. The response function for
HighSpeed TCP is
w ~.1/p^0.835 (7)
Comparing (6) and (7) we get k to be around 0.8. Since it's difficult
to implement an arbitrary power we choose k = 0.75 which can be
implemented using a fast integer algorithm for square root. Based on
extensive experimentation, we chose alpha = 1/8, beta = 1/2, and eta =
1. Substituting the above values for alpha, beta and k in (6) we get
the following response function for CTCP,
w = 0.255/p^0.8 (8)
The response function for CTCP is compared with HSTCP and is
illustrated in Table 1 below.
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CTCP HSTCP
Packet Drop Rate P Congestion Window W Congestion Window W
------------------ ------------------- -------------------
10^-3 64 38
10^-4 404 263
10^-5 2552 1795
10^-6 16107 12279
10^-7 101630 83981
10^-8 641245 574356
10^-9 4045987 3928088
10^-10 25528453 26864653
Table 1: TCP Response function for CTCP & HSTCP
The values in Table 1 illustrate that our choice of parameters makes
CTCP slightly more aggressive than HSTCP in moderate and low packet
loss rates but approaches HSTCP for larger windows. The reason we
choose to do this is because unlike HighSpeed TCP, CTCP's delay control
is capable of scaling back on detecting incipient congestion. As a
result, we expect CTCP to be more TCP friendly than HighSpeed TCP. We
show that this is in fact the case even under low buffering conditions
in the presence of high statistical multiplexing. The fairness
considerations and choice of gamma are detailed in Sections 5 and 6.
5. Automatic Selection of Gamma
To effectively detect early congestions, CTCP requires estimating the
backlogged packets at the bottleneck queue and compares this estimate
to a pre-defined threshold gamma. However, setting this threshold gamma
is particularly difficult for CTCP (and for many other similar delay-
based approaches) because gamma largely depends on the network
configuration and the number of concurrent flows that compete for the
same bottleneck link. Such flows are, unfortunately, unknown to end-
systems. Based on experimentation over varying conditions we originally
selected gamma to be 30 packets. This value appeared to provide a good
tradeoff between TCP fairness and throughput. However a fixed gamma can
still result in poor TCP friendliness over under-buffered network
links. One naive solution is to choose a very small value for gamma.
However this can falsely detect congestion and adversely affect
throughput. To address this problem, we instead use a method called
tuning-by-emulation to dynamically adjust gamma. The basic idea is to
estimate the backlogged packets of a Standard TCP flow along the same
path by simultaneously emulating the behavior of a Standard TCP flow.
Based on this, gamma is set so as to ensure good TCP-friendliness. CTCP
can then automatically adapt to different network configurations (i.e.,
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buffer provisioning) and also concurrent competing flows.
To ensure the effectiveness of incipient congestion detection, our
analytical model on CTCP shows that gamma should at least be less than
B/(m+l), where B is the bottleneck buffer and m and l represent the
number of concurrent Standard TCP flows and CTCP flows, respectively,
that are competing for the same bottleneck link [CTCPI06][CTCPP06]
[CTCPT]. Generally, both B and (m+l) are unknown to end-systems. It is
very difficult to estimate these values from end-systems in real-time,
especially the number of flows, which can vary significantly over time.
Fortunately there is a way to directly estimate the ratio B/(m+l), even
though the individual variables B and (m+l) are hard to estimate. Let's
first assume there are (m+l) regular TCP flows in the network. These
(m+l) flows should be able to fairly share the bottleneck capacity in
steady state. Therefore, they should also get roughly equal shares of
the buffers at the bottleneck, which should equal to B/(m+l). For such
a Standard TCP flow, although it does not know either B or (m+l), it
can still infer B/(m+l) easily by estimating its backlogged packets,
which is a rather mature technique widely used in many delay-based
protocols. This brings us to the core idea of CTCP's algorithm; CTCP
lets the sender emulate the congestion window of a Standard TCP flow.
Using this emulated window, we can estimate the buffer occupancy
(diff_reno) for a Standard TCP flow. Diff_reno can be regarded as a
conservative estimate of B/(m+l) assuming that the high speed flow is
more aggressive than Standard TCP. By choosing gamma <= diff_reno, we
can ensure TCP fairness.
The implementation is actually fairly trivial. This is because CTCP
already emulates Standard TCP as the loss-based component. We can
simply estimate the buffer occupancy of a competing Standard TCP flow
from state that CTCP already maintains. We choose an initial gamma = 30
and diff_reno is calculated as follows,
expected_reno (throughput) = cwnd/basertt
actual_reno (throughput) = cwnd/srtt
diff_reno = (expected - actual) * basertt
The difference between diff_reno and diff is simply that diff_reno is
computed only using the loss-based component cwnd. Since Standard TCP
reaches its maximum buffer occupancy just before a loss, CTCP uses the
diff_reno value computed in the previous round to calculate the gamma
for the next round. Whenever a loss happens, gamma is chosen to be less
than diff_reno and the sample values of gamma are updated using a
standard exponentially weighted moving average. The pseudocode to
calculate gamma is shown below. Here a round tracks every window's
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worth of data. Section 7 provides more details on how to maintain a
round.
Initialization:
diff_reno = invalid;
Gamma = 30;
End-of-Round:
expected_reno = cwnd / baseRTT;
actual_reno = cwnd / RTT;
diff_reno = (Expected_reno-Actual_reno)*baseRTT;
On-Packet-Loss:
If diff_reno is valid then
g_sample = 3/4*Diff_reno;
gamma = gamma*(1-lamda)+ lamda*g_sample;
if (gamma < gamma_low)
gamma=gamma_low;
else if (gamma > gamma_high)
gamma=gamma_high;
fi
diff_reno = invalid;
fi
The recommended values for gamma_low and gamma_high are 5 and 30
respectively. diff_reno is set to invalid to prevent using stale
diff_reno data when there are consecutive losses between which no
samples were taken.
6. Implementation Issues
CTCP has been implemented on Microsoft Windows and there has been
extensive testing on production links and in Windows Beta deployments.
The first challenge is to design a mechanism that can precisely track
the changes in round trip time with minimal overhead, and can scale
well to support many concurrent TCP connections. Naively taking RTT
samples for every packet will obviously be an over-kill for both CPU
and system memory, especially for high-speed and long distance networks
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where the congestion window can be very large. Therefore, CTCP needs to
limit the number of samples taken, but without compromising on
accuracy. In our implementation, we only take up to M samples per
window of data. M is chosen to scale with the round trip delay and
window size.
In order to further improve the efficiency in memory usage, we have
developed a memory allocation mechanism to dynamically allocate sample
buffers from a kernel fixed-size per-processor pool. The size should be
chosen as a function of the available system memory. As the window size
increases, M can be updated so that the samples are uniformly
distributed over the window. As M gets updated, more memory blocks are
allocated and linked to the existing sample buffers. If the sending
rate changes, either due to network conditions or due to application
behavior, the sample blocks are reclaimed to the global memory pool.
This dynamic buffer management ensures the scalability of our
implementation, so that it can work well even in a busy server which
could host tens of thousands of TCP connections simultaneously. Note
that it may also require a high-resolution timer to time RTT samples.
The rest of the implementation is rather straightforward. We add two
new state variables into the standard TCP Control Block, namely dwnd
and basertt (described in Section 3). Following the common practice of
high-speed protocols, CTCP reverts to standard TCP behavior when the
window is small. Delay-based component only kicks in when cwnd is
larger than some threshold, currently set to 38 packets assuming 1500
byte MTU. dwnd is updated at the end of each round. Note that no RTT
sampling and dwnd update happens during the loss recovery phase. This
is because the retransmission during the loss recovery phase may result
in inaccurate RTT samples and can adversely affect the delay-based
control.
7. Deployment Issues
There are several variations of TCP proposed for high speed and long
delay networks. We do not claim Compound TCP to be the best nor the
most optimal algorithm. However, based on extensive testing via
simulations and experimentation including those on production links as
well as beta deployments of a reasonable scale, we believe that
Compound TCP satisfies the design considerations outlined earlier in
this document. It effectively uses spare bandwidth in high speed
networks, achieves good intra-protocol fairness even in the presence of
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differing RTTs and does not adversely impact standard TCP. Furthermore,
Compound TCP does not require any changes or any new feedback from the
network and is deployable over the current Internet in an incremental
fashion. It interoperates with Standard TCP and requires support only
on the send side of a TCP connection for it to be used.
We also note that similar to High Speed TCP, in environments typical of
much of the current Internet, Compound TCP behaves exactly like
Standard TCP. This it does by ensuring that it follows the standard TCP
algorithm without any modification any time the congestion window is
less than 38 packets. Only when the congestion window is greater than
38 packets does the delay-based component of Compound TCP get invoked.
Thus, for example for a connection with an RTT of 100ms, the end-to-end
bandwidth must be greater than 4.8Mbps for CTCP to have any difference
in its response to network conditions compared to standard TCP.
Further, we do not believe that the deployment of Compound TCP would
block the possible deployment of alternate experimental congestion
control algorithms such as Fast TCP [FAST] or CUBIC [CUBIC]. In
particular, Compound TCP's response has a fallback to a loss-based
function that has characteristics very similar to HS-TCP or N parallel
TCP connections.
8. Security Considerations
This proposal makes no changes to the underlying security of the TCP
protocol.
9. IANA Considerations
There are no IANA considerations regarding this proposal.
10. Conclusions
This document proposes a congestion control algorithm for TCP for high
speed and long delay networks. By introducing a delay-based component
in addition to a standard TCP-based loss component, Compound TCP is
able to detect and effectively use spare bandwidth that may be
available on a high speed and long delay network. Furthermore, the
delay-based component detects the onset of congestion early and
gracefully reduces the sending rate. The loss-based component, on the
other hand, ensures there is an effective response to losses in network
while in the absence of losses, keeps the throughput of CTCP lower
bounded by TCP Reno. Thus, CTCP is not timid, nor does it induce more
self-induced packet loss than a single standard TCP flow. Thus Compound
TCP is efficient in consuming available bandwidth while being friendly
to standard TCP. Further, the delay component does not have any RTT
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bias thereby reducing the RTT bias of the Compound TCP vis-a-vis
standard TCP.
Compound TCP has been implemented as an optional component in Microsoft
Windows Vista. It has been tested and experimented through broad
Windows Vista beta deployments where it has been verified to meet its
objectives without causing any adverse impact. The Stanford Linear
Accelerator Center (SLAC) has also evaluated Compound TCP on production
links. Based on testing and evaluation done so far, we believe Compound
TCP is safe to deploy on the current Internet. We welcome additional
analysis, testing and evaluation of Compound TCP by Internet community
at large and continue to do additional testing ourselves.
11. Acknowledgments
The authors would like to thank Jingmin Song for all his efforts in
evaluating the algorithm on the test beds. We are thankful to Yee-ting
Lee and Les Cottrell for testing and evaluation of Compound TCP on
Internet2 links [SLAC]. We would like to thank Sanjay Kaniyar for his
insightful comments and for driving this project in Microsoft. We are
also thankful to the Microsft.com data center staff who helped us
evaluate Compound TCP on their production links. In addition, several
folks from the Internet research community who attended the High-Speed
TCP Summit at Microsoft [MSWRK] have provided valuable feedback on
Compound TCP. Finally, we are thankful to the Windows Vista program
beta participants who helped us test and evaluate CTCP.
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12. References
12.1. Normative References
[CTCPI06] K. Tan, Jingmin Song, Qian Zhang, Murari Sridharan, "A
Compound TCP Approach for High-speed and Long Distance
Networks", in IEEE Infocom, April 2006, Barcelona, Spain.
[RFC2581] Allman, M., Paxson, V. and W. Stevens, "TCP Congestion
Control", RFC 2581, April 1999.
12.2. Informative References
[AFRICA] R. King, R. Baraniuk and R. riedi, "TCP-Africa: An
Adaptive and Fair Rapid Increase Rule for Scalable
TCP", In Proc. INFOCOM 2005.
[BAINF01] Bansal and H. Balakrishnan, "Binomial Congestion Control
Algorithms", Proc INFOCOM 2001.
[CTCPP06] K. Tan, J. Song, Q. Zhang, and M. Sridharan, "Compound
TCP: A Scalable and TCP-friendly Congestion Control
for High-speed Networks", in 4th International
workshop on Protocols for Fast Long-Distance Networks
(PFLDNet), 2006, Nara, Japan.
[CTCPT] K. Tan, J. Song, M. Sridharan, and C.Y. Ho, "CTCP:
Improving TCP-Friendliness Over Low-Buffered Network
Links", Microsoft Technical Report.
[CUBIC] I. Rhee, L. Xu and S. Ha, "CUBIC for fast long
distance networks", Internet Draft, Expires Aug 31,
2007, draft-rhee-tcp-cubic-00.txt
[FAST] C. Jin, D. Wei, S. Low, "FAST TCP: Motivation,
Architecture, Algorithms, Performance", in IEEE Infocom
2004.
[MSWRK] Microsoft High-Speed TCP Summit,
http://research.microsoft.com/events/TCPSummit/
[PADHYE] J. Padhya, V. Firoiu, D. Towsley and J. Kurose,
"Modeling TCP Throughput: A Simple Model and its
Empirical Validation", in Proc. ACM SIGCOMM 1998.
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[RFC2988] V. Paxon and M. Allman, "Computing TCP's Retransmission
Timer", RFC 2988, November 2000.
[RFC3649] S. Floyd, "HighSpeed TCP for Large Congestion
Windows", RFC 3649, Dec 2003.
[SLAC] Yee-Ting Li, "Evaluation of TCP Congestion Control
Algorithms on the Windows Vista Platform", SLAC-TN-06-
005, http://www.slac.stanford.edu/pubs/slactns/tn04/slac-
tn-06-005.pdf
[VEGAS] L. Brakmo, S. O'Malley, and L. Peterson, "TCP Vegas:
New techniques for congestion detection and
avoidance", in Proc. ACM SIGCOMM, 1994.
Author's Addresses
Murari Sridharan
Microsoft Corporation
1 Microsoft Way, Redmond 98052
Email: muraris@microsoft.com
Kun Tan
Microsoft Research
5/F, Beijing Sigma Center
No.49, Zhichun Road, Hai Dian District
Beijing China 100080
Email: kuntan@microsoft.com
Deepak Bansal
Microsoft Corporation
1 Microsoft Way, Redmond 98052
Email: dbansal@microsoft.com
Dave Thaler
Microsoft Corporation
1 Microsoft Way, Redmond 98052
Email: dthaler@microsoft.com
Sridharan Expires April 29, 2008 [Page 16]
Internet-Draft Compound TCP October 2007
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