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TCP Details: Roadmap
 Congestion Control: Causes, Symptoms,
Approaches to Dealing With
 Slow Start/ Congestion Avoidance
 TCP Fairness
 TCP Performance
 Transport Layer Wrap-up
3: Transport Layer
3b-1
Principles of Congestion Control
Congestion:
 informally: “too many sources sending too much
data too fast for network to handle”
 different from flow control!
 a top-10 problem!
3: Transport Layer
3b-2
Congestion Signals
 Lost packets:If there are more packets
than resources (ex. Buffer space) along
some path, then no choice but to drop some
 Delayed packets: Router queues get full
and packets wait longer for service
 Explicit notification: Routers can alter
packet headers to notify end hosts
3: Transport Layer
3b-3
Congestion Collapse
 As number of packets entering network
increases, number of packets arriving at
destination increases but only up to a point
 Packet dropped in network => all the
resources it used along the way are wasted
=> no forward progress
 Internet 1987
3: Transport Layer
3b-4
Congestion Prevention?
 In a connection-oriented network:

Prevent congestion by requiring resources to be
reserved in advance
 In a connectionless network:
 No
prevention for congestion, just reaction to
congestion (congestion control)
3: Transport Layer
3b-5
Causes/costs of congestion: scenario 1
 two senders, two
receivers
 one router,
infinite buffers
 no retransmission
 large delays
when congested
 maximum
achievable
throughput
3: Transport Layer
3b-6
Causes/costs of congestion: scenario 2
 one router, finite buffers
 sender retransmission of lost packet
3: Transport Layer
3b-7
Causes/costs of congestion: scenario 2

l
=
l out (goodput)
in
 “perfect” retransmission only when loss:

l > lout
in
retransmission of delayed (not lost) packet makes l
in
l
(than perfect case) for same
out
larger
“costs” of congestion:
 more work (retrans) for given “goodput”
 unneeded retransmissions: link carries multiple copies of pkt
3: Transport Layer
3b-8
Causes/costs of congestion: scenario 3
 four senders
 multihop paths
 timeout/retransmit
Q: what happens as l
in
and l increase ?
in
3: Transport Layer
3b-9
Causes/costs of congestion: scenario 3
Another “cost” of congestion:
 when packet dropped, any “upstream transmission
capacity used for that packet was wasted!
3: Transport Layer 3b-10
Approaches towards congestion control
Two broad approaches towards congestion control:
End-end congestion
control:
 no explicit feedback from
network
 congestion inferred from
end-system observed loss,
delay
 approach taken by TCP
Network-assisted
congestion control:
 routers provide feedback
to end systems
 single bit indicating
congestion (SNA,
DECbit, TCP/IP ECN,
ATM)
 explicit rate sender
should send at
3: Transport Layer 3b-11
Window Size Revised
 Limit window size by both receiver
advertised window *and* a “congestion
window”
MaxWindow < = minimum (ReceiverAdvertised
Window, Congestion Window)
 EffectiveWindow = Max Window - (Last
ByteSent - LastByteAcked)

3: Transport Layer 3b-12
TCP Congestion Control
 end-end control (no network assistance)
 transmission rate limited by congestion window size, Congwin,
over segments:
Congwin
3: Transport Layer 3b-13
Original: With Just Flow
Control
Destination
…
Source
3: Transport Layer 3b-14
TCP Congestion Control: Two
Phases
 two “phases”
slow start
 congestion avoidance

 important variables:
 Congwin: current congestion window
 Threshold: defines threshold between two
slow start phase, congestion control phase
3: Transport Layer 3b-15
TCP congestion control:
 “probing” for usable
bandwidth:



ideally: transmit as fast
as possible (Congwin as
large as possible)
without loss
increase Congwin until
loss (congestion)
loss: decrease Congwin,
then begin probing
(increasing) again
 Don’t just send the entire
receiver’s advertised
window worth of data right
away
 Start with a congestion
window of 1 or 2 packets
 Slow start: For each ack
received, double window up
until a threshold, then just
increase by 1
 Congestion Avoidance: For
each timeout, start back at
1 and halve the upper
threshold
3: Transport Layer 3b-16
“Slow” Start:
Multiplicative Increase
Source
Destination
Multiplicative Increase Up to the Threshold
“Slower” than full receiver’s advertised
window
…
Faster than additive increase
3: Transport Layer 3b-17
TCP Congestion Avoidance:
Additive Increase
Source
Destination
…
Additive Increase Past the Threshhold
3: Transport Layer 3b-18
TCP Congestion Avoidance:
Multiplicative Decrease too
Congestion avoidance
/* slowstart is over
*/
/* Congwin > threshold */
Until (loss event) {
every w segments ACKed:
Congwin++
}
threshold = Congwin/2
Congwin = 1
1
perform slowstart
1: TCP Reno skips slowstart (fast
recovery) after three duplicate ACKs
3: Transport Layer 3b-19
Fast Retransmit
 Interpret 3 duplicate acks as an early
warning of loss (other causes? Reordering
or duplication in network)
 As if timeout - Retransmit packet and set
the slow-start threshold to half the
amount of unacked data
 Unlike timeout - set congestion window to
the threshhold (not back to 1 like normal
slow start)
3: Transport Layer 3b-20
Fast Recovery
 After a fast retransmit, do congestion
avoidance but not slow start.
 After third dup ack received:
threshold = ½ (congestion window)
 Congestion window = threshold + 2* MSS

 If more dup acks arrive:
 congestion Window += MSS
 When ack arrives for new data,deflate
congestion window:

congestionWindow = threshold
3: Transport Layer 3b-21
KB
Connection Timeline
70
60
50
40
30
20
10
1.0
2.0
3.0
4.0
5.0
6.0
7.0
8.0
9.0
Time (seconds)





blue line = value of congestion window in KB
Short hash marks = segment transmission
Long hash lines = time when a packet eventually
retransmitted was first transmitted
Dot at top of graph = timeout
0-0.4 Slow start; 2.0 timeout, start back at 1; 2.0-4.0 linear
increase
3: Transport Layer 3b-22
AIMD
TCP congestion
avoidance:
 AIMD: additive
increase,
multiplicative
decrease


increase window by 1
per RTT
decrease window by
factor of 2 on loss
event
TCP Fairness
Fairness goal: if N TCP
sessions share same
bottleneck link, each
should get 1/N of link
capacity
TCP connection 1
TCP
connection 2
bottleneck
router
capacity R
3: Transport Layer 3b-23
Why is TCP fair?
Two competing sessions:
 Additive increase gives slope of 1, as throughout increases
 multiplicative decrease decreases throughput proportionally

R
equal bandwidth share
loss: decrease window by factor of 2
congestion avoidance: additive increase
loss: decrease window by factor of 2
congestion avoidance: additive increase
Connection 1 throughput R
3: Transport Layer 3b-24
TCP Congestion Control History
 Before 1988, only flow control!
 TCP Tahoe 1988
 Congestion control with multiplicative decrease on
timeout
 TCP Reno 1990
 Add fast recovery and delayed acknowledgements
 TCP Vegas ?
 Tries
to use space in router’s queues fairly not
just divide BW fairly
3: Transport Layer 3b-25
TCP Vegas
 Tries to use constant space in the router
buffers
 Compares each round trip time to the
minimum round trip time it has seen to
infer time spent in queuing delays
 Vegas in not a recommended version of TCP
Minimum time may never happen
 Can’t compete with Tahoe or Reno

3: Transport Layer 3b-26
TCP latency modeling
Q: How long does it take to receive an object from a
Web server after sending a request?
 TCP connection establishment
 data transfer delay
 Slow start
A: That is a natural question, but not very easy to
answer. Depends on round trip time, bandwidth,
window size (dynamic changes to it)
3: Transport Layer 3b-27
TCP latency modeling
Two cases to consider:
 Slow Sender (Big Window): Still sending when ACK returns


time to send window
W*S/R
> time to get first ack
> RTT + S/R
 Fast Sender (Small Window):Wait for ACK to send more data


time to send window
W*S/R
< time to get first ack
< RTT + S/R
Notation, assumptions:
 O: object size (bits)
 Assume one link between client and server of rate R
 Assume: fixed congestion window, W segments
 S: MSS (bits)
 no retransmissions (no loss, no corruption)
3: Transport Layer 3b-28
TCP latency Modeling
Slow Sender (Big Window):
latency = 2RTT + O/R
Number of windows:
K := O/WS
Fast Sender (Small Window):
latency = 2RTT + O/R
+ (K-1)[S/R + RTT - WS/R]
(S/R + RTT) – (WS/R) = Time Till Ack Arrives –
Time to Transmit Window 3: Transport Layer
3b-29
TCP Latency Modeling: Slow Start
 Now suppose window grows according to slow start (not slow
start + congestion avoidance).
 Will show that the latency of one object of size O is:
Latency  2 RTT 
O
S
S

 P  RTT    ( 2 P  1)
R
R
R

where P is the number of times TCP stalls at server waiting
for Ack to arrive and open the window:
P  min {Q, K  1}
- Q is the number of times the server would stall
if the object were of infinite size - maybe 0.
- K is the number of windows that cover the object.
-S/R is time to transmit one segment
- RTT+ S/R is time to get ACK of one segment
3: Transport Layer 3b-30
TCP Latency Modeling: Slow Start (cont.)
Example:
O/S = 15 segments
K = 4 windows
initiate TCP
connection
request
object
Stall 1
first window
= S/R
RTT
second window
= 2S/R
Q=2
Stall 2
third window
= 4S/R
P = min{K-1,Q} = 2
Server stalls P=2 times.
fourth window
= 8S/R
complete
transmission
object
delivered
time at
client
time at
server
3: Transport Layer 3b-31
TCP Latency Modeling: Slow Start (cont.)
S
 RTT  time from when server starts to send segment
R
until server receives acknowledg ement
initiate TCP
connection
2k 1
S
 time to transmit the kth window
R

request
object
S
k 1 S 

RTT

2
 stall time after the kth window
 R
R 
first window
= S/R
RTT
second window
= 2S/R
third window
= 4S/R
P
O
latency   2 RTT   stallTime p
R
p 1
P
O
S
S
  2 RTT   [  RTT  2k 1 ]
R
R
k 1 R
O
S
S
  2 RTT  P[ RTT  ]  ( 2 P  1)
R
R
R
fourth window
= 8S/R
complete
transmission
object
delivered
time at
client
time at
server
3: Transport Layer 3b-32
TCP Performance Limits
 Can’t go faster than speed of slowest link
between sender and receiver
 Can’t go faster than
receiverAdvertisedWindow/RoundTripTime
 Can’t go faster than 2*RTT
 Can’t go faster than memory bandwidth
(lost of memory copies in the kernel)
3: Transport Layer 3b-33
Experiment: Compare TCP and
UDP performance
 Use ttcp (or pcattcp) to compare effective
BW when transmitting the same size data
over TCP and UDP
 UDP not limited by overheads from
connection setup or flow control or
congestion control
 Use Ethereal to trace both
3: Transport Layer 3b-34
TCP vs UDP
What would happen if UDP used more than TCP?
3: Transport Layer 3b-35
Transport Layer Summary
 principles behind
transport layer services:
multiplexing/demultiplexing
 reliable data transfer
 flow control
 congestion control
 instantiation and
implementation in the Internet
 UDP
 TCP

Next:
 leaving the network
“edge” (application
transport layer)
 into the network “core”
3: Transport Layer 3b-36
Outtakes
3: Transport Layer 3b-37
In-order Delivery
 Each packet contains a sequence number
 TCP layer will not deliver any packet to the
application unless it has already received
and delivered all previous messages
 Held in receive buffer
3: Transport Layer 3b-38
Sliding Window Protocol
 Reliable Delivery - by acknowledgments and
retransmission
 In-order Delivery - by sequence number
 Flow Control - by window size
 These properites guaranteed end-to-end
not per-hop
3: Transport Layer 3b-39
Segment Transmission
 Maximum segment size reached
If accumulate MSS worth of data, send
 MSS usually set to MTU of the directly
connected network (minus TCP/IP headers)

 Sender explicitly requests
 If sender requests a push, send
 Periodic timer
 If data held for too long, sent
3: Transport Layer 3b-40
 1)
To aid in congestion control, when a packet is
dropped the Timeout is set tp double the last
Timeout. Suppose a TCP connection, with window
size 1, loses every other packet. Those that do
arrive have RTT= 1 second. What happens? What
happens to TimeOut? Do this for two cases:

 a.
After a packet is eventually received, we pick
up where we left off, resuming EstimatedRTT
initialized to its pretimeout value and Timeout
double that as usual.
 b.
After a packet is eventually received, we
resume with TimeOut initialized to the last
exponentially backed-off value used for the
3: Transport Layer
timeout interval.
3b-41
Case study: ATM ABR congestion control
ABR: available bit rate:
 “elastic service”
 if sender’s path
“underloaded”:
 sender should use
available bandwidth
 if sender’s path
congested:
 sender throttled to
minimum guaranteed
rate
RM (resource management)
cells:
 sent by sender, interspersed
with data cells
 bits in RM cell set by switches
(“network-assisted”)
 NI bit: no increase in rate
(mild congestion)
 CI bit: congestion
indication
 RM cells returned to sender by
receiver, with bits intact
3: Transport Layer 3b-42
Case study: ATM ABR congestion control
 two-byte ER (explicit rate) field in RM cell
 congested switch may lower ER value in cell
 sender’ send rate thus minimum supportable rate on path
 EFCI bit in data cells: set to 1 in congested switch
 if data cell preceding RM cell has EFCI set, sender sets CI
bit in returned RM cell
3: Transport Layer 3b-43
Sliding Window Protocol
 Reliable Delivery - by acknowledgments and
retransmission
 In-order Delivery - by sequence number
 Flow Control - by window size
 These properites guaranteed end-to-end
not per-hop
3: Transport Layer 3b-44
End to End Argument
 TCP must guarantee reliability, in-order,
flow control end-to-end even if guaranteed
for each step along way - why?
Packets may take different paths through
network
 Packets pass through intermediates that might
be misbehaving

3: Transport Layer 3b-45
End-To-End Arguement
 A function should not be provided in the
lower levels unless it can be completely and
correctly implemented at that level.
 Lower levels may implement functions as
performance optimization. CRC on hop to
hop basis because detecting and
retransmitting a single corrupt packet
across one hop avoid retransmitting
everything end-to-end
3: Transport Layer 3b-46
TCP vs sliding window on
physical, point-to-point link
 1) Unlike physical link, need connection
establishment/termination to setup or tear
down the logical link
 2) Round-trip times can vary significantly
over lifetime of connection due to delay in
network so need adaptive retransmission
timer
 3) Packets can be reordered in Internet
(not possible on point-to-point)
3: Transport Layer 3b-47
TCP vs point-to-point
(continues)
 4) establish maximum segment lifetime
based on IP time-to-live field conservative estimate of how the TTL field
(hops) translates into MSL (time)
 5) On point-to-point link can assume
computer on each end have enough buffer
space to support the link

TCP must learn buffering on other end
3: Transport Layer 3b-48
TCP vs point-to-point
(continued)
 6) no congestion on a point-to-point link -
TCP fast sender could swamp slow link on
route to receiver or multiple senders could
swamp a link on path
 need
congestion control in TCP
3: Transport Layer 3b-49