Advanced Networks Transport layer: TCP
Dr. Wei Bao | Lecturer School of Computer Science
Pipelined protocols: overview
Go-back-N:
›sender can have up to N unacked packets in pipeline
› receiver only sends cumulative ack
– does not ack packet if there is a gap
› sender has timer for oldest unacked packet
– when timer expires, retransmit all unacked packets
Selective Repeat:
›sender can have up to N unacked packets in pipeline
› receiver sends individual ack for each packet
› sender maintains timer for each unacked packet
– when timer expires, retransmit only that unacked packet
2
Go-Back-N: sender
› “window” of up to N, consecutive unacked pkts allowed
v ACK(n): ACKs all pkts up to, including seq # n – “cumulative ACK”
§ may receive duplicate ACKs (see receiver)
v timer for oldest in-flight pkt
v timeout(n): retransmit packet n and all higher seq # pkts in window
3
Go-Back-N: sender
› “window” of up to N, consecutive unacked pkts allowed
v ACK(n): ACKs all pkts up to, including seq # n – “cumulative ACK”
§ may receive duplicate ACKs (see receiver)
v timer for oldest in-flight pkt
v timeout(n): retransmit packet n and all higher seq # pkts in window
4
GBN: sender extended FSM
rdt_send(data)
if (nextseqnum < base+N) {
sndpkt[nextseqnum] = make_pkt(nextseqnum,data,chksum) udt_send(sndpkt[nextseqnum])
if (base == nextseqnum)
L
base=1 nextseqnum=1
rdt_rcv(rcvpkt)
&& corrupt(rcvpkt)
start_timer nextseqnum++
} else refuse_data(data)
Wait
timeout
start_timer udt_send(sndpkt[base]) udt_send(sndpkt[base+1])
... udt_send(sndpkt[nextseqnum-1])
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt)
base = getacknum(rcvpkt)+1 If (base == nextseqnum)
stop_timer else
start_timer
3-5
GBN: receiver extended FSM
L
expectedseqnum=1
default udt_send(sndpkt)
Wait
rdt_rcv(rcvpkt)
&& notcurrupt(rcvpkt)
&& hasseqnum(rcvpkt,expectedseqnum)
extract(rcvpkt,data)
deliver_data(data)
sndpkt = make_pkt(expectedseqnum,ACK,chksum) udt_send(sndpkt)
expectedseqnum++
ACK-only: always send ACK for correctly-received pkt with highest in-order seq #
- may generate duplicate ACKs
- need only remember expectedseqnum
› out-of-order pkt:
- discard (don’t buffer): no receiver buffering! - re-ACK pkt with highest in-order seq #
6
GBN in action
sender window (N=4)
012345678 012345678 45678 45678
012345678 01 2345678
sender
receiver
receive pkt0, send ack0 receive pkt1, send ack1
receive pkt3, discard, (re)send ack1
receive pkt4, discard,
(re)send ack1 receive pkt5, discard,
(re)send ack1
rcv pkt2, deliver, send ack2 rcv pkt3, deliver, send ack3 rcv pkt4, deliver, send ack4 rcv pkt5, deliver, send ack5
send pkt0 send pkt1 send pkt2 send pkt3
0123
0123
2345
2345
2345
2345
01 678 01 678 01 678 01 678
rcv ack0, send pkt4 rcv ack1, send pkt5
ignore duplicate ACK
pkt 2 timeout
send pkt2 send pkt3 send pkt4 send pkt5
Xloss (wait)
3-7
Selective repeat
› receiver individually acknowledges all correctly received pkts
- buffers pkts, as needed, for eventual in-order delivery to upper layer
› sender only resends pkts for which ACK not received
- sender timer for each unACKed pkt
› sender window
› receiver window
8
Selective repeat: sender, receiver windows
3-9
Selective repeat
sender
data from above:
› if next available seq # in window, send pkt
timeout(n):
› resend pkt n, restart timer
ACK(n) in [sendbase,sendbase+N-1]:
› mark pkt n as received
› if n is smallest unACKed pkt, advance window base to next unACKed seq #
receiver
pkt n in [rcvbase, rcvbase+N-1]
v send ACK(n)
v out-of-order: buffer
v in-order: deliver (also deliver buffered, in-order pkts), advance window to next not-yet-received pkt
pkt n in [rcvbase-N,rcvbase-1]
v ACK(n) otherwise:
v ignore
10
3-11
Selective repeat in action
sender window (N=4)
012345678 012345678 45678 45678
012345678 012345678
012345678
01 678 01 678 01 678
012345678 0123456789
sender
receiver
receive pkt0, send ack0 receive pkt1, send ack1
receive pkt3, buffer, send ack3
receive pkt4, buffer,
send ack4 receive pkt5, buffer,
send ack5
rcv pkt2; deliver pkt2, pkt3, pkt4, pkt5; send ack2
send pkt0 send pkt1 send pkt2 send pkt3
0123
0123
Xloss (wait)
rcv ack0, send pkt4 rcv ack1, send pkt5
record ack3 arrived
pkt 2 timeout
send pkt2
record ack4 arrived record ack5 arrived
Q: what happens when ack2 arrives?
2345
2345
2345
3-12
Connection-oriented Transport TCP
13
TCP: Overview RFCs: 793,1122,1323, 2018, 2581
› point-to-point:
- one sender, one receiver
› reliable, in-order byte stream › pipelined:
- TCP congestion and flow control set window size
› full duplex data:
- bi-directional data flow in
same connection
- MSS: maximum segment size
› connection-oriented:
- handshaking (exchange of control msgs) inits sender, receiver state before data exchange
› flow controlled:
- sender will not overwhelm receiver
14
TCP segment structure
URG: urgent data (generally not used)
ACK: ACK # valid
PSH: push data now (generally not used)
RST, SYN, FIN: connection estab (setup, teardown commands)
Internet checksum (as in UDP)
counting
by bytes
of data
(not segments!)
# bytes rcvr willing to accept
32 bits
source port #
sequence number
acknowledgement number
dest port #
head len
not used
checksum
U
A
P
R
S
F
options (variable length)
application
data
(variable length)
receive window
Urg data pointer
15
TCP seq. numbers, ACKs
sequence numbers:
- “number” of first byte in segment’s data
acknowledgements:
- seq # of next byte expected from other side
- cumulative ACK
Q: how receiver handles out-of-
order segments
- A: TCP spec doesn’t say,
- up to implementor
- Most will store, but still use cumulative ACK
outgoing segment from sender
source port #
dest port #
sequence number
acknowledgement number
rwnd
checksum
urg pointer
window size N
sender sequence number space
sent ACKed
sent, not- yet ACKed (“in- flight”)
usable not but not usable yet sent
incoming segment to sender
source port #
dest port #
sequence number
acknowledgement number
A
rwnd
checksum
urg pointer
16
TCP seq. numbers, ACKs
Host A Host B
User types ‘C’
host ACKs receipt
of echoed ‘C’
Seq=4201, ACK=7901, data = ‘C’, 4201- 4300
Seq=7901, ACK=4301, data = ‘C’ 7901-8000
host ACKs receipt of ‘C’, echoes back ‘C’
Seq=4301, ACK=8001 no data
simple telnet scenario
17
TCP round trip time, timeout
Q: how to set TCP timeout value?
› longer than RTT - but RTT varies
› too short: premature timeout, unnecessary retransmissions
› too long: slow reaction to segment loss
Q: how to estimate RTT?
› SampleRTT: measured time from segment transmission until ACK receipt
- ignore retransmissions
› SampleRTT will vary, want
estimated RTT “smoother”
- weighted average of several recent measurements, not just current SampleRTT
18
SampleRT
Host A Host B
SampleRTT
SampleRTT
19
Ignore retransmissions
Host A
Host B
Host A
Host B
First attempt
First attempt
SampleRTT?
Retransmission ACK
SampleRTT?
Retransmission ACK
SampleRTT?
SampleRTT?
20
TCP round trip time, timeout
EstimatedRTT = (1- a)*EstimatedRTT + a*SampleRTT
v exponential weighted moving average
v influence of past sample decreases exponentially fast
v typical value: a = 0.125
RTT: gaia.cs.umass.edu to fantasia.eurecom.fr
350
300
250
200
150
100
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
RTT: gaia.cs.umass.edu to fantasia.eurecom.fr
sampleRTT
EstimatedRTT
time (seconnds)
time (seconds)
SampleRTT Estimated RTT 21
RTT (milliseconds)
RTT (milliseconds)
TCP round trip time, timeout
› timeout interval: EstimatedRTT plus “safety margin” - large variation in EstimatedRTT -> larger safety margin
› estimate SampleRTT deviation from EstimatedRTT: DevRTT = (1-b)*DevRTT +
b*|SampleRTT-EstimatedRTT| (typically, b = 0.25)
TimeoutInterval = EstimatedRTT + 4*DevRTT
estimated RTT
“safety margin”
22
Reliable Data Transfer in TCP
23
TCP reliable data transfer
› TCP creates rdt service on top of IP’s unreliable service
– pipelined segments
– cumulative acks
– single retransmission timer
›retransmissions triggered by:
– timeout events – duplicate acks
let’s initially consider simplified TCP sender:
– ignore duplicate acks
– ignore flow control, congestion control
24
TCP sender events
data rcvd from app:
› create segment with seq #
› seq # is byte-stream number of first data byte in segment
› start timer if not already running
– think of timer as for oldest unacked segment
– expiration interval: TimeOutInterval
timeout:
› retransmit segment that caused timeout
› restart timer ack rcvd:
› if ack acknowledges previously unacked segments
– update what is known to be ACKed
– start timer if there are still unacked segments
25
TCP sender (simplified)
L
NextSeqNum = InitialSeqNum SendBase = InitialSeqNum
wait for event
data received from application above
create segment, seq. #: NextSeqNum
pass segment to IP (i.e., “send”) NextSeqNum = NextSeqNum + length(data) if (timer currently not running)
start timer
timeout
retransmit not-yet-acked segment with smallest seq. #
start timer
ACK received, with ACK field value y
if (y > SendBase) {
SendBase = y
/* SendBase–1: last cumulatively ACKed byte */ if (there are currently not-yet-acked segments)
start timer else stop timer
}
26
TCP: retransmission scenarios
Host A Host B
Seq=92, 8 bytes of data
Host A Host B SendBase=92
ACK=100
X
Seq=92, 8 bytes of data
ACK=100
lost ACK scenario
SendBase=100 SendBase=120
SendBase=120
Seq=92, 8 bytes of data Seq=100, 20 bytes of data
ACK=100 ACK=120
Seq=92, 8 bytes of data
ACK=120
premature timeout
27
timeout
timeout
TCP: retransmission scenarios
Host A
Host B
Seq=92, 8 bytes of data
Seq=100, 20 bytes of data
X ACK=100 ACK=120
Seq=120, 15 bytes of data
cumulative ACK
28
timeout
TCP ACK generation [RFC 1122, RFC 2581]
event at receiver
arrival of in-order segment with expected seq #. All data up to expected seq # already ACKed
arrival of in-order segment with expected seq #. One other segment has ACK pending
arrival of out-of-order segment higher-than-expect seq. # . Gap detected
arrival of segment that partially or completely fills gap
TCP receiver action
delayed ACK. Wait up to 500ms
for next segment. If no next segment, send ACK
immediately send single cumulative ACK, ACKing both in-order segments
immediately send duplicate ACK, indicating seq. # of next expected byte
immediate send ACK, provided that segment starts at lower end of gap
29
TCP ACK
Host A
Host B
Host A
Host B
Too many ACKs
cumulative ACK
30
TCP ACK
Host A Host B Host A Host B
Need to Yes ack this
one?
31
TCP fast retransmit
› time-out period often relatively long:
– long delay before resending lost packet
› detect lost segments via duplicate ACKs.
– sender often sends many segments back-to-back
– if segment is lost, there will likely be many duplicate ACKs.
TCP fast retransmit
if sender receives 3 duplicate ACKs for same data
(“triple duplicate ACKs”), resend unacked segment with smallest seq #
§ likely that unacked segment lost, so don’t wait for timeout
32
TCP fast retransmit
Host A Host B
Seq=92, 8 bytes of data
Seq=100, 20 bytes of data
X
ACK=100
ACK=100
ACK=100 ACK=100
Seq=100, 20 bytes of data
fast retransmit after sender receipt of triple duplicate ACK
33
timeout
Flow Control in TCP
34
TCP flow control
application process
TCP code
TCP socket receiver buffers
IP code
application may remove data from TCP socket buffers ….
… slower than TCP receiver is delivering (sender is sending)
application OS
flow control
receiver controls sender, so sender won’t overflow receiver’s buffer by transmitting too much, too fast
from sender
receiver protocol stack
35
TCP flow control
› receiver “advertises” free buffer space by including rwnd value in TCP header of receiver-to- sender segments
– RcvBuffer size set via socket options (typical default is 4096
bytes)
– many operating systems autoadjust
RcvBuffer
› sender limits amount of unacked (“in-flight”) data to receiver’s rwnd value
› guarantees receive buffer will not overflow
to application process
buffered data
free buffer space
RcvBuffer
rwnd
TCP segment payloads receiver-side buffering
36
Connection Management in TCP
37
Connection Management
before exchanging data, sender/receiver “handshake”:
› agree to establish connection (each knowing the other willing to establish connection)
› agree on connection parameters
application
connection state: ESTAB connection variables:
seq # client-to-server server-to-client
rcvBuffer size at server,client
network
application
connection state: ESTAB connection Variables:
seq # client-to-server server-to-client
rcvBuffer size at server,client
network
38
Agreeing to establish a connection
2-way handshake:
Let’s talk OK
ESTAB
Q: will 2-way handshake always work in network?
› variable delays
› retransmitted messages (e.g.
req_conn(x)) due to message loss
› message reordering
› can’t “see” other side
ESTAB
choose x
ESTAB
req_conn(x)
ESTAB
acc_conn(x)
39
Agreeing to establish a connection
2-way handshake failure scenarios:
choose x
retransmit req_conn(x)
ESTAB
client terminates
req_conn(x)
ESTAB
choose x
retransmit req_conn(x)
ESTAB
retransmit data(x+1)
client terminates
req_conn(x)
ESTAB
accept data(x+1)
server forgets x
ESTAB
accept data(x+1)
acc_conn(x) req_conn(x)
connection x completes
server forgets x
ESTAB
acc_conn(x) data(x+1)
connection x completes
req_conn(x) data(x+1)
half open connection! (no client!)
40
TCP 3-way handshake
client state
LISTEN
choose init seq num, x
server state
LISTEN
SYN RCVD
SYNSENT
SYNbit=1, Seq=x
SYNbit=1, Seq=y ACKbit=1; ACKnum=x+1
ACKbit=1, ACKnum=y+1
choose init seq num, y send TCP SYNACK msg, acking SYN
send TCP SYN msg
ESTAB
received SYNACK(x) indicates server is live; send ACK for SYNACK;
this segment may contain client-to-server data
received ACK(y) indicates client is live
ESTAB
41
TCP: closing a connection
› client, server each closes their side of connection – send TCP segment with FIN bit = 1
› respond to received FIN with ACK
42
TCP: closing a connection
client state
server state
ESTAB
CLOSE_WAIT
LAST_ACK
CLOSED
ESTAB
CLOSED
clientSocket.close()
FIN_WAIT_1
FIN_WAIT_2
TIMED_WAIT
can no longer send but can receive data
wait for server close
FINbit=1, seq=x
ACKbit=1; ACKnum=x+1
FINbit=1, seq=y ACKbit=1; ACKnum=y+1
can still send data
timed wait for 2*max
segment lifetime
can no longer send data
43
TCP segment structure
URG: urgent data (generally not used)
ACK: ACK # valid
PSH: push data now (generally not used)
RST, SYN, FIN: connection estab (setup, teardown commands)
Internet checksum (as in UDP)
counting
by bytes
of data
(not segments!)
# bytes rcvr willing to accept
32 bits
source port #
sequence number
acknowledgement number
dest port #
head len
not used
checksum
U
A
P
R
S
F
options (variable length)
application
data
(variable length)
receive window
Urg data pointer
44
Principles of Congestion Control
45
Principles of congestion control
congestion:
› informally: “too many sources sending too much
data too fast for network to handle” › different from flow control!
› manifestations:
– lost packets (buffer overflow at routers)
– long delays (queueing in router buffers) › a top-10 problem!
46
Causes/costs of congestion: scenario 1
original data: lin › two senders, two receivers
Host A
throughput: lout unlimited shared
output link buffers
throughput: lout
› one router, infinite buffers
› output link capacity: R
› no retransmission original data: lin
Host B
R/2
lin R/2
lin R/2
v large delays as arrival rate, lin, approaches capacity
› maximum per-connection throughput: R/2
47
lout
delay
Causes/costs of congestion: scenario 2
› one router, finite buffers
› sender retransmission of timed-out packet
– application-layer input = application-layer output: lin = lout、 – Goodput
– transport-layer input includes retransmissions : l’in lin lin : original data
lout
l’in: original data, plus retransmitted data
Host A
Host B
finite shared output link buffers
48
Causes/costs of congestion: scenario 2
idealization: perfect knowledge
› sender sends only when router buffers available
R/2
copy
lin : original data
l’in: original data, plus
retransmitted data
free buffer space!
lout
l’in
R/2
A
Host B
finite shared output link buffers
49
lout
Causes/costs of congestion: scenario 2
Idealization: known loss packets can be lost, dropped at router due to full buffers
› sender only resends if packet known to be lost
copy
Host B
lin : original data lout l’in: original data, plus
retransmitted data
no buffer space!
A
50
Causes/costs of congestion: scenario 2
Idealization: known loss packets can be lost, dropped at router due to full buffers
› sender only resends if packet known to be lost
R/2
lin lin : original data
l’in: original data, plus retransmitted data
free buffer space!
when sending at R/2, some packets are retransmissions but asymptotic goodput is still R/2
R/2
lout
A
Host B
51
lout
Causes/costs of congestion: scenario 2
Realistic: duplicates
v packets can be lost, dropped
at router due to full buffers
v sender times out prematurely, sending two copies, both of which are delivered
R/2
lin
when sending at R/2, some packets are retransmissions including duplicated that are delivered!
R/2
l
timeout
lin l’in
copy
out
A
free buffer space!
Host B
52
lout
Causes/costs of congestion: scenario 2
Realistic: duplicates
v packets can be lost, dropped
at router due to full buffers
v sender times out prematurely, sending two copies, both of which are delivered
“costs” of congestion:
R/2
v more work (retrans) for given “goodput”
v unneeded retransmissions: link carries multiple copies of pkt
§ decreasing goodput
lin
R/2
when sending at R/2, some packets are retransmissions including duplicated that are delivered!
53
lout
Causes/costs of congestion: scenario 3
› four senders
› multihop paths
› timeout/retransmit
Q: what happens as lin’ increases ?
A: as red lin’ increases, all arriving blue pkts at upper queue are dropped, blue throughput g 0
lout l’in: original data, plus
Host A
lin : original data retransmitted data
Host B
finite shared output link buffers
Host D
Host C
54
Causes/costs of congestion: scenario 3
lin’
another “cost” of congestion:
v when packet dropped, any “upstream transmission capacity used for that packet was wasted!
55
lout
Approaches towards congestion control
two broad approaches towards congestion control:
end-end congestion network-assisted
control:
› no explicit feedback from network
› congestion inferred from end-system observed loss, delay
› approach taken by TCP
congestion control:
› routers provide feedback to end systems
– single bit indicating congestion
– explicit rate for sender to send at
56
TCP Congestion Control
57
TCP congestion control
Additive increase multiplicative decrease (AIMD)
v approach: sender increases transmission rate (window size), probing for usable bandwidth, until loss occurs
§additive increase: increase cwnd by 1 MSS (maximum segment size) every RTT until loss detected
§ multiplicative decrease: cut cwnd in half after loss additively increase window size …
…. until loss occurs (then cut window in half)
AIMD saw tooth behavior: probing for bandwidth
time
58
cwnd: TCP sender congestion window size
TCP Congestion Control: details
sender sequence number space
TCP sending rate:
› roughly: send cwnd bytes, wait RTT for ACKS, then send more bytes
cwnd
last byte ACKed
sent, not- yet ACKed (“in- flight”)
last byte sent
› sender limits transmission:
› cwnd is dynamic, function of perceived network congestion
rate ~
cwnd RTT
bytes/sec
LastByteSent-LastByteAcked < cwnd
59
TCP Slow Start
› when connection begins, increase rate exponentially:
- initially cwnd = 1 MSS
- double cwnd every RTT
- done by incrementing cwnd for every ACK received
› summary: initial rate is slow but ramps up exponentially fast
› when should the exponential increase switch to linear (additive increase)?
Host A
Host B
time
60
RTT
one segment
two segments
four segments
TCP: switching from slow start to CA
Q: when should the exponential increase switch to linear?
A: cwnd reaches ssthresh
ssthresh=6
loss!
cwnd=12
Implementation:
› Atbeginningssthresh,specifiedin different versions of TCP
› (Inthisexamplessthresh=8 segment)
› onlossevent,ssthreshissetto1/2 of cwnd just before loss event
61
TCP: detecting, reacting to loss
› loss indicated by timeout:
- cwnd set to 1 MSS;
- window then grows exponentially (as in slow start) to ssthresh, then grows linearly
› loss indicated by 3 duplicate ACKs:
› TCP Tahoe, same as loss indicated by timeout, always sets cwnd to 1 (timeout
or 3 duplicate acks) › TCP RENO
- cwnd is cut in half window then grows linearly (additive increase) - fast recovery
62
TCP: switching from slow start to CA
slow start
additive increase
additive increase multiplicative decease
additive increase slow start
63
Summary: TCP Reno Congestion Control
New ACK!
New ACK! .
duplicate ACK dupACKcount++
slow
start/ exponential increase
new ACK
new ACK
cwnd = cwnd + MSS (MSS/cwnd) dupACKcount = 0
transmit new segment(s), as allowed
L
cwnd = 1 MSS ssthresh = 64 KB dupACKcount = 0
timeout
ssthresh = cwnd/2
cwnd = 1 MSS dupACKcount = 0 retransmit missing segment
dupACKcount == 3
ssthresh= cwnd/2 cwnd = ssthresh + 3 retransmit missing segment
cwnd = cwnd+MSS
dupACKcount = 0
transmit new segment(s), as allowed
cwnd > ssthresh
L
timeout ssthresh = cwnd/2
cwnd = 1 MSS dupACKcount = 0 retransmit missing segment
congestion avoidance/ additive increase
duplicate ACK dupACKcount++
timeout
ssthresh = cwnd/2
cwnd = 1
dupACKcount = 0 retransmit missing segment
fast recovery/ multiplicative decrease
New ACK!
New ACK
cwnd = ssthresh dupACKcount = 0
dupACKcount == 3
ssthresh= cwnd/2
cwnd = ssthresh + 3 retransmit missing segment
duplicate ACK
cwnd = cwnd + MSS
transmit new segment(s), as allowed
64
TCP Reno throughput
› avg. TCP thruput as function of window size, RTT? – ignore slow start, assume always data to send
› W: window size (measured in bytes) where loss occurs – avg. window size (# in-flight bytes) is 3⁄4 W
– avg. thruput is 3/4W per RTT
avg TCP thruput = 3 W bytes/sec 4 RTT
W W/2
65
TCP Futures: TCP over “long, fat pipes”
› example: 1500 byte segments, 100ms RTT, want 10 Gbps throughput
› requires W = 83,333 in-flight segments
› throughput in terms of segment loss probability, L [Mathis
1997]:
TCP throughput = 1.22 . MSS RTT L
➜ to achieve 10 Gbps throughput, need a loss rate of L = 2·10-10 – a very small loss rate!
› new versions of TCP for high-speed › Vegas, Westwood, CUBIC, etc.
66
TCP Fairness
Fairness: K TCP sessions share same bottleneck link of bandwidth R, each has average rate of R/K
TCP connection 1
TCP connection 2
bottleneck router capacity R
67
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
45o
Session 1 data rate
R
68
Session 2 data rate
Fairness (more)
Fairness and UDP
› multimedia apps often do not use TCP
– do not want rate throttled by congestion control
› instead use UDP:
– send audio/video at constant rate, tolerate packet loss
Fairness, parallel TCP connections
› application can open multiple parallel connections between two hosts
› e.g., link of rate R
– App 1 asks for 1 TCP, gets 0.1R – App 2 asks for 9 TCPs, gets 0.9R
69
Final word
Window size = min (rwnd, cwnd)
congestion window flow control congestion control
receive window
70
Conclusion
› principles behind transport layer services: – multiplexing, demultiplexing
– reliable data transfer
– connection setup, teardown
– flow control
– congestion control
› instantiation, implementation in the Internet – UDP
– TCP
71