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End to End Protocols
We already saw: basic protocols Stop & wait (Correct but low performance)
Now: Window based protocol.
• Go Back N• Selective Repeat
TCP protocol. UDP protocol.
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Pipelined protocols
Pipelining: sender allows multiple, “in-flight”, yet-to-be-acknowledged pkts range of sequence numbers must be increased buffering at sender and/or receiver
Two generic forms of pipelined protocols: go-Back-N, selective repeat
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Go-Back-NSender: k-bit seq # in pkt header “window” of up to N, consecutive unack’ed pkts allowed
ACK(n): ACKs all pkts up to, including seq # n - “cumulative ACK” may deceive duplicate ACKs (see receiver)
timer for each in-flight pkt timeout(n): retransmit pkt n and all higher seq # pkts in window
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GBN: receiver extended FSM
receiver simple: 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! ACK pkt with highest in-order seq #
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GBN - correctness
Safety: The sequence
numbers guarantee: packet received in
order. No gaps. No duplicates. Safety follows from
extectedsequencenum Next seg. Received exactly once.
Liveness: Eventually timeout. Re-sends the window. Eventually base is
received correctly.
Receiver: from that time ACK at
least base. Eventually an ACK will
get through. The sender will
update to Base (or more).
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GBN - correctness
Clearing a FIFO channel:
Data k
Ack k
impossible
Data i<k
Ack i<k
impossible
Claim: After receiving Data/ACK k no Data/ACK i<k is received.
Sufficient to use N+1 seq. num.
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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 N consecutive seq #’s again limits seq #s of sent, unACKed pkts
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Selective repeat
data from above : if next available seq # in
window, send pkt
timeout(n): resend pkt n, restart
timer
ACK(n) in [sendbase,sendbase+N]:
mark pkt n as received if n smallest unACKed
pkt, advance window base to next unACKed seq #
senderpkt n in [rcvbase, rcvbase+N-
1]
send ACK(n) out-of-order: buffer in-order: deliver (also
deliver buffered, in-order pkts), advance window to next not-yet-received pkt
pkt n in [rcvbase-N,rcvbase-1]
ACK(n)
otherwise: ignore
receiver
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Selective Repeat - Correctness Infinite seq. Num.
Safety: immediate from the seq. Num. Liveness: Eventually data and ACKs get
through. Finite Seq. Num.
Idea: Re-use seq. Num. Use less bits to encode them.
Number of seq. Num.: At least N. Needs more!
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Selective repeat: dilemma
Example: seq #’s: 0, 1, 2, 3 window size=3
receiver sees no difference in two scenarios!
incorrectly passes duplicate data as new in (a)
Q: what relationship between seq # size and window size?
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Choosing the window size
Small window size: idle link (under-utilization).
Large window size: Buffer space Delay after loss
Ideal window size (assuming very low loss) RTT =Round trip time C = link capacity window size = RTT * C
What happens with no loss?
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applicationtransportnetwork
MP2
applicationtransportnetwork
Multiplexing/demultiplexing
Recall: segment - unit of data exchanged between transport
layer entities aka TPDU: transport
protocol data unitreceiver
HtHn
Demultiplexing: delivering received segments (TPDUs)to correct app layer processes
segment
segment Mapplicationtransportnetwork
P1M
M MP3 P4
segmentheader
application-layerdata
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Multiplexing/demultiplexing
multiplexing/demultiplexing: based on sender, receiver port
numbers, IP addresses source, dest port #s in each
segment recall: well-known port
numbers for specific applications
gathering data from multiple app processes, enveloping data with header (later used for demultiplexing)
source port # dest port #
32 bits
applicationdata
(message)
other header fields
TCP/UDP segment format
Multiplexing:
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Multiplexing/demultiplexing: examples
host A server Bsource port: xdest. port: 23
source port:23dest. port: x
port use: simple telnet app
WWW clienthost A
WWWserver B
WWW clienthost C
Source IP: CDest IP: B
source port: x
dest. port: 80
Source IP: CDest IP: B
source port: y
dest. port: 80
port use: WWW server
Source IP: ADest IP: B
source port: x
dest. port: 80
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TCP: Overview RFCs: 793, 1122, 1323, 2018, 2581
full duplex data: bi-directional data flow in
same connection MSS: maximum segment
size
connection-oriented: handshaking (exchange of
control msgs) init’s sender, receiver state before data exchange
flow controlled: sender will not overwhelm
receiver
point-to-point: one sender, one receiver
reliable, in-order byte steam: no “message boundaries”
pipelined: TCP congestion and flow
control set window size
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TCP segment structure
source port # dest port #
32 bits
applicationdata
(variable length)
sequence number
acknowledgement numberrcvr window size
ptr urgent datachecksum
FSRPAUheadlen
notused
Options (variable length)
URG: urgent data (generally not used)
ACK: ACK #valid
PSH: push data now(generally not used)
RST, SYN, FIN:connection estab(setup, teardown
commands)
# bytes rcvr willingto accept
countingby bytes of data(not segments!)
Internetchecksum
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TCP seq. #’s and ACKsSeq. #’s:
byte stream “number” of first byte in segment’s data
ACKs: 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
Host A Host B
Seq=42, ACK=79, data = ‘C’
Seq=79, ACK=43, data = ‘C’
Seq=43, ACK=80
Usertypes
‘C’
host ACKsreceipt
of echoed‘C’
host ACKsreceipt of
‘C’, echoesback ‘C’
timesimple telnet scenario
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TCP: reliable data transfer
simplified sender, assuming
waitfor
event
waitfor
event
event: data received from application above
event: timer timeout for segment with seq # y
event: ACK received,with ACK # y
create, send segment
retransmit segment
ACK processing
•one way data transfer•no flow, congestion control
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TCP: reliable data transfer
00 sendbase = initial_sequence number 01 nextseqnum = initial_sequence number 0203 loop (forever) { 04 switch(event) 05 event: data received from application above 06 create TCP segment with sequence number nextseqnum 07 start timer for segment nextseqnum 08 pass segment to IP 09 nextseqnum = nextseqnum + length(data) 10 event: timer timeout for segment with sequence number y 11 retransmit segment with sequence number y 12 compue new timeout interval for segment y 13 restart timer for sequence number y 14 event: ACK received, with ACK field value of y 15 if (y > sendbase) { /* cumulative ACK of all data up to y */ 16 cancel all timers for segments with sequence numbers < y 17 sendbase = y 18 } 19 else { /* a duplicate ACK for already ACKed segment */ 20 increment number of duplicate ACKs received for y 21 if (number of duplicate ACKS received for y == 3) { 22 /* TCP fast retransmit */ 23 resend segment with sequence number y 24 restart timer for segment y 25 } 26 } /* end of loop forever */
SimplifiedTCPsender
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TCP ACK generation [RFC 1122, RFC 2581]
Event
in-order segment arrival, no gaps,everything else already ACKed
in-order segment arrival, no gaps,one delayed ACK pending
out-of-order segment arrivalhigher-than-expect seq. #gap detected
arrival of segment that partially or completely fills gap
TCP Receiver action
delayed ACK. Wait up to 500msfor next segment. If no next segment,send ACK
immediately send singlecumulative ACK
send duplicate ACK, indicating seq. #of next expected byte
immediate ACK if segment startsat lower end of gap
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TCP: retransmission scenarios
Host A
Seq=92, 8 bytes data
ACK=100
loss
tim
eout
time lost ACK scenario
Host B
X
Seq=92, 8 bytes data
ACK=100
Host A
Seq=100, 20 bytes data
ACK=100
Seq=
92
tim
eout
time premature timeout,cumulative ACKs
Host B
Seq=92, 8 bytes data
ACK=120
Seq=92, 8 bytes data
Seq=
10
0 t
imeou
t
ACK=120
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TCP Flow Controlreceiver: explicitly
informs sender of (dynamically changing) amount of free buffer space RcvWindow field
in TCP segmentsender: keeps the
amount of transmitted, unACKed data less than most recently received RcvWindow
sender won’t overrun
receiver’s buffers bytransmitting too
much, too fast
flow control
receiver buffering
RcvBuffer = size or TCP Receive Buffer
RcvWindow = amount of spare room in Buffer
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TCP Round Trip Time and TimeoutQ: how to set TCP
timeout value? longer than RTT
note: RTT will vary 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,
cumulatively ACKed segments
SampleRTT will vary, want estimated RTT “smoother” use several recent
measurements, not just current SampleRTT
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TCP Round Trip Time and TimeoutEstimatedRTT = (1-x)*EstimatedRTT + x*SampleRTT
Exponential weighted moving average influence of given sample decreases exponentially fast typical value of x: 0.1
Setting the timeout EstimtedRTT plus “safety margin” large variation in EstimatedRTT -> larger safety margin
Timeout = EstimatedRTT + 4*Deviation
Deviation = (1-x)*Deviation + x*|SampleRTT-EstimatedRTT|
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TCP Connection Management
Recall: TCP sender, receiver establish “connection” before exchanging data segments
initialize TCP variables: seq. #s buffers, flow control info
(e.g. RcvWindow) client: connection initiator Socket clientSocket = new
Socket("hostname","port number"); server: contacted by client Socket connectionSocket =
welcomeSocket.accept();
Three way handshake:
Step 1: client sends TCP SYN control segment to server specifies initial seq #
Step 2: server receives SYN, replies with SYNACK control segment
ACKs received SYN allocates buffers specifies server-to-receiver
initial seq. #
Step 3: client sends ACK and data.
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TCP Connection Management (cont.)
Closing a connection:
client closes socket: clientSocket.close();
Step 1: client end system sends TCP FIN control segment to server.
Step 2: server receives FIN, replies with ACK. Closes connection, sends FIN.
client
FIN
server
ACK
ACK
FIN
close
close
closed
tim
ed w
ait
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TCP Connection Management (cont.)
Step 3: client receives FIN, replies with ACK.
Enters “timed wait” - will respond with ACK to received FINs
Step 4: server, receives ACK. Connection closed.
Note: with small modification, can handly simultaneous FINs.
client
FIN
server
ACK
ACK
FIN
closing
closing
closed
tim
ed w
ait
closed
38
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!
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Causes/costs of congestion: scenario 1
two senders, two receivers
one router, infinite buffers
no retransmission
large delays when congested
maximum achievable throughput
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Causes/costs of congestion: scenario 2
one router, finite buffers sender retransmission of lost packet
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Causes/costs of congestion: scenario 2 always: (goodput)
“perfect” retransmission only when loss:
retransmission of delayed (not lost) packet makes
larger (than perfect case) for same
in
out
=
in
out
>
in
out
“costs” of congestion: more work (retrans) for given “goodput” unneeded retransmissions: link carries multiple copies of pkt
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Causes/costs of congestion: scenario 3 four senders multihop paths timeout/retransmit
in
Q: what happens as and increase ?
in
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Causes/costs of congestion: scenario 3
Another “cost” of congestion: when packet dropped, any “upstream transmission capacity
used for that packet was wasted!
44
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
Two broad approaches towards congestion control:
45
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
46
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
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TCP Congestion Control end-end control (no network assistance) transmission rate limited by congestion window size, Congwin,
over segments:
w segments, each with MSS bytes sent in one RTT:
throughput = w * MSS
RTT Bytes/sec
Congwin
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TCP congestion control:
two “phases” slow start congestion avoidance
important variables: Congwin threshold: defines
threshold between two slow start phase, congestion control phase
“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
49
TCP Slowstart
exponential increase (per RTT) in window size (not so slow!)
loss event: timeout (Tahoe TCP) and/or or three duplicate ACKs (Reno TCP)
initialize: Congwin = 1for (each segment ACKed) Congwin++until (loss event OR CongWin > threshold)
Slowstart algorithmHost A
one segment
RTT
Host B
time
two segments
four segments
50
TCP Congestion Avoidance
/* slowstart is over */ /* Congwin > threshold */Until (loss event) { every w segments ACKed: Congwin++ }threshold = Congwin/2Congwin = 1perform slowstart
Congestion avoidance
1
1: TCP Reno skips slowstart (fast recovery) after three duplicate ACKs
Tahoe
Reno
51
TCP FairnessFairness goal: if N TCP
sessions share same bottleneck link, each should get 1/N of link capacity
TCP congestion avoidance:
AIMD: additive increase, multiplicative decrease increase window by
1 per RTT decrease window
by factor of 2 on loss event
AIMD
TCP connection 1
bottleneckrouter
capacity R
TCP connection 2
52
Why is TCP fair?
Two competing sessions: Additive increase gives slope of 1, as throughout increases multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughputConnect
ion 2
th
roughput
congestion avoidance: additive increaseloss: decrease window by factor of 2
congestion avoidance: additive increaseloss: decrease window by factor of 2
55
From Reno to Vegas
Reno Vegas
RTT measurem-t
coarse fine
Fast Retrans 3 dup ack Dup ack + timer
CongWin decrease
For each Fast Retrans.
For the 1st Fast Retrans. in RTT
Congestion detection
Loss detection Also based on measured thruput
57
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
Notation, assumptions: Assume one link between
client and server of rate R
Assume: fixed congestion window, W segments
S: MSS (bits) O: object size (bits) no retransmissions (no
loss, no corruption)Two cases to consider: WS/R > RTT + S/R: ACK for first segment in
window returns before window’s worth of data sent
WS/R < RTT + S/R: wait for ACK after sending window’s worth of data sent
58
TCP latency Modeling
Case 1: latency = 2RTT + O/R Case 2: latency = 2RTT + O/R+ (K-1)[S/R + RTT - WS/R]
K:= O/WS
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TCP Latency Modeling: Slow Start
Now suppose window grows according to slow start. Will show that the latency of one object of size O is:
R
S
R
SRTTP
R
ORTTLatency P )12(2
where P is the number of times TCP stalls at server:
}1,{min KQP
- where Q is the number of times the server would stall if the object were of infinite size.
- and K is the number of windows that cover the object.
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TCP Latency Modeling: Slow Start (cont.)
RTT
initia te TCPconnection
requestobject
first w indow= S /R
second w indow= 2S /R
third w indow= 4S /R
fourth w indow= 8S /R
com pletetransm issionobject
delivered
tim e atc lient
tim e atserver
Example:
O/S = 15 segments
K = 4 windows
Q = 2
P = min{K-1,Q} = 2
Server stalls P=2 times.
61
TCP Latency Modeling: Slow Start (cont.)
R
S
R
SRTTPRTT
R
O
R
SRTT
R
SRTT
R
O
stallTimeRTTR
O
P
kP
k
P
pp
)12(][2
]2[2
2latency
1
1
1
windowth after the timestall 2 1 kR
SRTT
R
S k
ementacknowledg receivesserver until
segment send tostartsserver whenfrom time RTTR
S
window kth the transmit totime2 1
R
Sk
RTT
initia te TCPconnection
requestobject
first w indow= S /R
second w indow= 2S /R
third w indow= 4S /R
fourth w indow= 8S /R
com pletetransm issionobject
delivered
tim e atc lient
tim e atserver
63
UDP: User Datagram Protocol [RFC 768]
“no frills,” “bare bones” Internet transport protocol
“best effort” service, UDP segments may be: lost delivered out of order to
app connectionless:
no handshaking between UDP sender, receiver
each UDP segment handled independently of others
Why is there a UDP? no connection establishment
(which can add delay) simple: no connection state at
sender, receiver small segment header no congestion control: UDP can
blast away as fast as desired
64
UDP: more
often used for streaming multimedia apps loss tolerant rate sensitive
other UDP uses (why?): DNS SNMP
reliable transfer over UDP: add reliability at application layer application-specific error
recover!
source port # dest port #
32 bits
Applicationdata
(message)
UDP segment format
length checksumLength, in
bytes of UDPsegment,including
header
65
UDP checksum
Sender: treat segment contents as
sequence of 16-bit integers checksum: addition (1’s
complement sum) of segment contents
sender puts checksum value into UDP checksum field
Receiver: compute checksum of received
segment check if computed checksum
equals checksum field value: NO - error detected YES - no error detected. But
maybe errors nonethless?
Goal: detect “errors” (e.g., flipped bits) in transmitted segment