Network Working Group M. Sridharan
Internet Draft Microsoft
Intended status: Experimental July 18, 2007 K. Tan
Expires: January 2008 Microsoft Research
D. Bansal
D. Thaler
Microsoft
Compound TCP: A New TCP Congestion Control for High-Speed and Long
Distance Networks
draft-sridharan-tcpm-ctcp-00.txt
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Abstract
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This document proposes Compound TCP (CTCP), a modification to TCP's
congestion control mechanism for use with TCP connections with large
congestion windows. 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.
We have implemented CTCP on Microsoft's Windows and we have done
extensive testing on production links and in Windows Beta
deployments. We also engaged with Stanford Linear Accelerator Center
to evaluate the properties of CTCP. The results so far are very
encouraging. This document describes the Compound TCP algorithm in
detail, and solicits experimentation and feedback from the wider
community. 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.
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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 ................................... 12
7. Deployment Issues........................................ 13
8. Security Considerations.................................. 13
9. IANA Considerations...................................... 13
10. Conclusions............................................. 14
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
This document proposes 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 will take 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
around 10^-11 with 1500-byte size packets (one packet loss over
2,600,000,000 packet transmission!), which is not practical in today's
networks.
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There are several proposals to address this fundamental limitation of
TCP. One straightforward way to overcome this limitation is to modify
TCP control'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 approaches 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]). The 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 to efficiency requirement and
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 a nice property of adjusting its
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 to the standard TCP's congestion avoidance algorithm.
Using the delay component as an automatic tuning knob, CTCP is scalable
yet TCP friendly.
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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
component in CTCP. Thus, the CTCP sending window now is controlled by
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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 the 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 sense 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. Our 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. The CTCP
sender also maintains a smoothed RTT srtt, updated as specified in
[RFC2988]. Then, the number of backlogged packets of the connection can
be estimated using,
expected (throughput) = wnd/basertt
actual (throughput) = wnd/srtt
diff = (expected - actual) * basertt
The expected throughput gives the estimation of throughput CTCP gets if
it does not overrun the network path. The actual throughput stands for
the throughput CTCP really gets. Using this we can calculate the amount
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of data backlogged in the bottleneck queue (diff). 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. We present an
effective way to automatically estimate gamma later in later sections.
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]. More specifically,
when no congestion is detected, CTCP window increases using the
following function
dwnd(t+1) = dwnd(t) + alpha*dwnd(t)^k (3)
When a packet loss occurs, the delay window is multiplicatively
decreased,
dwnd(t+1) = dwnd(t)*(1-beta) (4)
where alpha, beta and k are tunable to obtain the desirable
scalability, smoothness and responsiveness. We assume that a loss is
detected by three duplicate ACKs. 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, and the overall CTCP should follows the behavior
defined in (3) and (4). We now summarize the control law for CTCP's
delay component as follows;
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dwnd(t+1) =
dwnd(t) + alpha*dwnd(t)^k - 1, if diff < gamma (5)
dwnd(t) - eta*diff, if diff >= gamma (6)
dwnd(t)(1-beta) - cwnd/2, on packet loss (7)
where (5) 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, dwnd is set
to the difference between the desired reduced window size and that can
be provided by cwnd. The rule in equation (6) 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. It is because that 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)) (8)
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 (9)
Comparing (8) and (9) 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 choose alpha = 1/8 and beta = 1/2.
Substituting the above values for alpha, beta and k in (8) we get the
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following response function for CTCP,
w = 0.255/p^0.8 (10)
The response function for CTCP is compared with HSTCP and is
illustrated in Table 1 below.
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 later sections.
5. Automatic Selection of Gamma
To effectively detect early congestions, CTCP requires estimating the
backlogged packets at bottleneck queue and compares this estimate to a
pre-defined threshold gamma. However, setting this threshold gamma is
particular difficult for CTCP (and to 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, which are, unfortunately, unknown to end-systems. Based on
experimentation over varying conditions we selected gamma to be 30
packets. This value provided a pretty 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
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is to choose a very small value for gamma, however this can falsely
detect congestion and adversely affect throughput. To address this
problem we use a method called tuning-by-emulation to dynamically
adjust gamma. The basic idea of our proposal is to estimate the
backlogged packets of a Standard TCP flow along the same path by
emulating the behavior of a Standard TCP flow in runtime. Based on
this, gamma is set so as to ensure good TCP-friendliness. CTCP can then
automatically adapt to different network configurations (i.e., buffer
provisioning) and also concurrent competing flows.
Our analytical model on CTCP shows that gamma should at least be less
than B/m+l to ensure the effectiveness of incipient congestion
detection, where m and l present the flow number of concurrent Standard
TCP flows and CTCP flows 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 or
(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 share 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 (Q) for a Standard TCP flow. Q 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 <=
Q, we can ensure TCP fairness.
The implementation is actually 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 which CTCP already maintains. We choose an initial gamma = 30 and
Q is calculated as follows,
expected_reno (throughput) = cwnd/basertt
actual_reno (throughput) = cwnd/srtt
diff_reno = (expected - actual) * basertt
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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 earlier 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 worth
of data. We will provide more details on how to maintain a round in
Section 7.
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.
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6. Implementation Issues
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
sample for every packet will obviously be an over-kill for both CPU and
system memory, especially for high-speed and long distance networks
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 sample 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 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. The basertt is a value that tracks the minimum RTT sample
measured seen so far and it is used as an estimation of the
transmission delay of a single packet. Basertt is usually cleared if a
retransmission timeout is hit. It is a good idea to re-measure the
basertt incase the network conditions have changed. 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. It is because the retransmission during the loss recovery phase
may result in inaccurate RTT samples and can adversely affect the
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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 our extensive testing via
simulations, 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 before in
this document. It effectively uses spare bandwidth in high speed
networks, achieves good intra-protocol fairness even in the presence of
differing RTTs and does not adversely impact standard TCP. Further,
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 inter-operates with Standard TCP and requires support only
one 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 is follows standard TCP
algorithm without any modification any time congestion window is less
than 38 packets. Only when congestion window is greater than 38
packets, does the delay based component of Compound TCP gets invoked.
Thus, for example for a connection with RTT of 100ms, end to end
bandwidth must be greater than 4.8Mbps for CTCP algorithm to have any
difference in its response to network conditions than a 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 TCPs response has a fallback to 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.
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10. Conclusions
This document proposes a novel 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. Further, delay based
component detects onset of congestion early and gracefully reduces
sending rate. The loss based component, on the other hand, ensures
there is 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 induces 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 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 Operating System. 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. 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
[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] D. Bansal and H. Balakrishnan, "Binomial Congestion
Control Algorithms", Proc INFOCOM 2001.
[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.
[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. Paxson 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.
Authors' 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 January 18, 2008 [Page 16]
Internet-Draft Compound TCP July 2007
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