Title: Chapter 3: Transport Layer
1Chapter 3 Transport Layer
- learn about transport layer protocols in the
Internet - UDP connectionless transport
- TCP connection-oriented transport
- TCP congestion control
- Our goals
- understand principles behind transport layer
services - multiplexing/demultiplexing
- reliable data transfer
- flow control
- congestion control
2Chapter 3 outline
- 3.1 Transport-layer services
- 3.2 Multiplexing and demultiplexing
- 3.3 Connectionless transport UDP
- 3.4 Principles of reliable data transfer
- 3.5 Connection-oriented transport TCP
- segment structure
- reliable data transfer
- flow control
- connection management
- 3.6 Principles of congestion control
- 3.7 TCP congestion control
3Transport services and protocols
- provide logical communication between app
processes running on different hosts - transport protocols run in end systems
- sending side breaks app messages into segments,
passes to network layer - receiving side reassembles segments into
messages, passes to app layer - more than one transport protocol available to
apps - Internet TCP and UDP
4Transport vs. network layer
- network layer logical communication between
hosts - The Internets network layer provides a
best-effort, unreliable delivery service packets
may be lost, garbled, and out of order - transport layer logical communication between
processes - relies on, enhances, network layer services
5Internet transport-layer protocols
- reliable, connection-oriented, in-order delivery
TCP - congestion control
- flow control
- unreliable, connectionless, unordered delivery
UDP - services not available
- delay guarantees
- bandwidth guarantees
6Chapter 3 outline
- 3.1 Transport-layer services
- 3.2 Multiplexing and demultiplexing
- 3.3 Connectionless transport UDP
- 3.4 Principles of reliable data transfer
- 3.5 Connection-oriented transport TCP
- segment structure
- reliable data transfer
- flow control
- connection management
- 3.6 Principles of congestion control
- 3.7 TCP congestion control
7Multiplexing/demultiplexing
gathering data from different sockets, enveloping
data with header (later used for
demultiplexing), pass to network layer
8How demultiplexing works
32 bits
- each segment has source, destination port number
(recall well-known port numbers for specific
applications) - transport layer uses destination port number to
direct segment to appropriate socket
source port
dest port
other header fields
application data (message)
TCP/UDP segment format
9Connectionless demultiplexing
- UDP socket identified by two-tuple
- (dest IP address, dest port number)
- When host receives UDP segment
- checks destination port number in segment
- directs UDP segment to socket with that port
number - IP datagrams with different source IP addresses
and/or source port numbers directed to same
socket
10Connectionless demux (cont)
Source Port provides return address
11Connection-oriented demux
- TCP socket identified by 4-tuple
- source IP address
- source port number
- dest IP address
- dest port number
- recv host uses all four values to direct segment
to appropriate socket
- Web servers have different sockets for each
connecting client
12Connection-oriented demux (cont)
SP 9157
Client IPB
DP 80
server IP C
13Chapter 3 outline
- 3.1 Transport-layer services
- 3.2 Multiplexing and demultiplexing
- 3.3 Connectionless transport UDP
- 3.4 Principles of reliable data transfer
- 3.5 Connection-oriented transport TCP
- segment structure
- reliable data transfer
- flow control
- connection management
- 3.6 Principles of congestion control
- 3.7 TCP congestion control
14UDP User Datagram Protocol RFC 768
- no frills, bare bones Internet transport
protocol - multiplexing/demultiplexing
- error checking
- best effort service, UDP segments may be
- lost
- delivered out of order to app
- connectionless
- no handshaking between UDP sender, receiver
- 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
15UDP more
- often used for streaming multimedia apps
- loss tolerant
- rate sensitive
- other UDP uses
- DNS
- SNMP
- reliable transfer over UDP add reliability at
application layer - application-specific error recovery!
32 bits
source port
dest port
Length, in bytes of UDP segment, including header
checksum
length
Application data (message)
UDP segment format
16UDP checksum
- Goal detect errors (e.g., flipped bits) in
transmitted segment
- Sender
- treat segment contents as sequence of 16-bit
words - checksum 1s complement of 1s complement sum
(overflow from the MSB is added into the LSB) of
segment contents - sender puts checksum value into UDP checksum field
- Receiver
- compute sum of all 16-bit words of received
segment, including the checksum - if the sum contains all 1s
- NO - error detected
- YES - no error detected. But maybe errors
nonetheless
17Chapter 3 outline
- 3.1 Transport-layer services
- 3.2 Multiplexing and demultiplexing
- 3.3 Connectionless transport UDP
- 3.4 Principles of reliable data transfer
- 3.5 Connection-oriented transport TCP
- segment structure
- reliable data transfer
- flow control
- connection management
- 3.6 Principles of congestion control
- 3.7 TCP congestion control
18Principles of Reliable data transfer
- important in app., transport, link layers
- reliable channel no bit errors, no loss,
in-order delivery
- characteristics of unreliable channel will
determine complexity of reliable data transfer
protocol (rdt)
19Reliable data transfer getting started
send side
receive side
20Reliable data transfer getting started
- Well
- incrementally develop sender, receiver sides of
reliable data transfer protocol (rdt) - consider only unidirectional data transfer
- but control info will flow on both directions!
- use finite state machines (FSM) to specify
sender, receiver
event causing state transition
actions taken on state transition
initial state
state when in this state next state uniquely
determined by next event
21Rdt1.0 reliable transfer over a reliable channel
- underlying channel perfectly reliable
- no bit errors
- no loss of packets
- separate FSMs for sender, receiver
- sender sends data into underlying channel
- receiver reads data from underlying channel
rdt_send(data)
rdt_rcv(packet)
Wait for call from below
Wait for call from above
extract (packet,data) deliver_data(data)
packet make_pkt(data) udt_send(packet)
sender
receiver
22Rdt2.0 channel with bit errors
- underlying channel may flip bits in packet
- the question how to recover from errors
- error detection use checksum bits
- acknowledgements (ACKs) receiver explicitly
tells sender that pkt received OK - negative acknowledgements (NAKs) receiver
explicitly tells sender that pkt had errors - sender retransmits pkt on receipt of NAK
- new mechanisms in rdt2.0 (beyond rdt1.0)
- error detection
- receiver feedback control msgs (ACK,NAK)
rcvr-gtsender
23rdt2.0 FSM specification
rdt_send(data)
receiver
sndkpkt make_pkt(data, checksum) udt_send(sndpkt
)
rdt_rcv(rcvpkt) isNAK(rcvpkt)
Wait for call from above
udt_send(sndpkt)
rdt_rcv(rcvpkt) isACK(rcvpkt)
L
sender
rdt_rcv(rcvpkt) notcorrupt(rcvpkt)
Sender sends one packet, then waits for receiver
response
extract(rcvpkt,data) deliver_data(data) udt_send(A
CK)
24rdt2.0 operation with no errors
rdt_send(data)
sndpkt make_pkt(data, checksum) udt_send(sndpkt)
rdt_rcv(rcvpkt) isNAK(rcvpkt)
Wait for call from above
udt_send(sndpkt)
rdt_rcv(rcvpkt) isACK(rcvpkt)
Wait for call from below
L
rdt_rcv(rcvpkt) notcorrupt(rcvpkt)
extract(rcvpkt,data) deliver_data(data) udt_send(A
CK)
25rdt2.0 error scenario
rdt_send(data)
sndpkt make_pkt(data, checksum) udt_send(sndpkt)
rdt_rcv(rcvpkt) isNAK(rcvpkt)
Wait for call from above
udt_send(sndpkt)
rdt_rcv(rcvpkt) isACK(rcvpkt)
Wait for call from below
L
rdt_rcv(rcvpkt) notcorrupt(rcvpkt)
extract(rcvpkt,data) deliver_data(data) udt_send(A
CK)
26rdt2.0 has a fatal flaw!
- What happens if ACK/NAK corrupted?
- Add checksum bits to ACK/NACK
- sender doesnt know what happened at receiver!
- What to do?
- sender ACKs/NAKs receivers ACK/NAK? What if
sender ACK/NAK corrupted? - retransmit, but this might cause retransmission
of correctly received pkt!
- Handling duplicates
- sender adds sequence number to each pkt
- sender retransmits current pkt if ACK/NAK garbled
- receiver discards (doesnt deliver up) duplicate
pkt - A one-bit sequence number will suffice
27rdt2.1 sender, handles garbled ACK/NAKs
rdt_send(data)
sndpkt make_pkt(0, data, checksum) udt_send(sndp
kt)
rdt_rcv(rcvpkt) ( corrupt(rcvpkt)
isNAK(rcvpkt) )
Wait for call 0 from above
udt_send(sndpkt)
rdt_rcv(rcvpkt) notcorrupt(rcvpkt)
isACK(rcvpkt)
rdt_rcv(rcvpkt) notcorrupt(rcvpkt)
isACK(rcvpkt)
L
L
rdt_rcv(rcvpkt) ( corrupt(rcvpkt)
isNAK(rcvpkt) )
rdt_send(data)
sndpkt make_pkt(1, data, checksum) udt_send(sndp
kt)
udt_send(sndpkt)
28rdt2.1 receiver, handles garbled ACK/NAKs
rdt_rcv(rcvpkt) notcorrupt(rcvpkt)
has_seq0(rcvpkt)
extract(rcvpkt,data) deliver_data(data) sndpkt
make_pkt(ACK, chksum) udt_send(sndpkt)
rdt_rcv(rcvpkt) (corrupt(rcvpkt)
rdt_rcv(rcvpkt) (corrupt(rcvpkt)
sndpkt make_pkt(NAK, chksum) udt_send(sndpkt)
sndpkt make_pkt(NAK, chksum) udt_send(sndpkt)
rdt_rcv(rcvpkt) not corrupt(rcvpkt)
has_seq1(rcvpkt)
rdt_rcv(rcvpkt) not corrupt(rcvpkt)
has_seq0(rcvpkt)
sndpkt make_pkt(ACK, chksum) udt_send(sndpkt)
sndpkt make_pkt(ACK, chksum) udt_send(sndpkt)
rdt_rcv(rcvpkt) notcorrupt(rcvpkt)
has_seq1(rcvpkt)
extract(rcvpkt,data) deliver_data(data) sndpkt
make_pkt(ACK, chksum) udt_send(sndpkt)
29rdt2.1 discussion
- Sender
- 1-bit seq added to pkt
- must check if received ACK/NAK corrupted
- twice as many states
- state must remember whether current pkt has 0
or 1 seq.
- Receiver
- must check if received packet is duplicate
- state indicates whether 0 or 1 is expected pkt
seq
30rdt2.2 a NAK-free protocol
- same functionality as rdt2.1, using ACKs only
- instead of NAK, receiver sends ACK for last pkt
received OK - receiver must explicitly include seq of pkt
being ACKed - duplicate ACK at sender results in same action as
NAK retransmit current pkt
31rdt2.2 sender, receiver fragments
rdt_send(data)
sndpkt make_pkt(0, data, checksum) udt_send(sndp
kt)
rdt_rcv(rcvpkt) ( corrupt(rcvpkt)
isACK(rcvpkt,1) )
udt_send(sndpkt)
sender FSM fragment
rdt_rcv(rcvpkt) notcorrupt(rcvpkt)
isACK(rcvpkt,0)
rdt_rcv(rcvpkt) (corrupt(rcvpkt)
has_seq1(rcvpkt))
L
receiver FSM fragment
sndpkt make_pkt (ACK, 1, chksum) udt_send(sndpkt
)
rdt_rcv(rcvpkt) notcorrupt(rcvpkt)
has_seq1(rcvpkt)
extract(rcvpkt,data) deliver_data(data) sndpkt
make_pkt(ACK, 1, chksum) udt_send(sndpkt)
32rdt3.0 channels with errors and loss
- New assumption underlying channel can also lose
packets (data or ACKs) - Q how to deal with loss?
- sender waits until certain data or ACK lost, then
retransmits
- Approach sender waits reasonable amount of
time for ACK - retransmits if no ACK received in this time
- if pkt (or ACK) just delayed (not lost)
- retransmission will be duplicate, but use of
seq. s already handles this - requires countdown timer
33rdt3.0 sender
rdt_send(data)
rdt_rcv(rcvpkt) ( corrupt(rcvpkt)
isACK(rcvpkt,1) )
sndpkt make_pkt(0, data, checksum) udt_send(sndp
kt) start_timer
L
rdt_rcv(rcvpkt)
L
timeout
udt_send(sndpkt) start_timer
rdt_rcv(rcvpkt) notcorrupt(rcvpkt)
isACK(rcvpkt,1)
rdt_rcv(rcvpkt) notcorrupt(rcvpkt)
isACK(rcvpkt,0)
stop_timer
stop_timer
timeout
udt_send(sndpkt) start_timer
rdt_rcv(rcvpkt)
L
rdt_send(data)
rdt_rcv(rcvpkt) ( corrupt(rcvpkt)
isACK(rcvpkt,0) )
sndpkt make_pkt(1, data, checksum) udt_send(sndp
kt) start_timer
L
34rdt3.0 in action
35rdt3.0 in action
36Performance of rdt3.0
- rdt3.0 works, but performance stinks
- example 1 Gbps link, 15 ms e-e prop. delay, 1KB
packet
L (packet length in bits)
8kb/pkt
T
0.008 ms
transmit
R (transmission rate, bps)
109 b/sec
- U sender utilization fraction of time sender
busy sending - 1KB pkt every 30.008 msec -gt 267kbps throughput
over 1 Gbps link - network protocol limits use of physical resources!
37rdt3.0 stop-and-wait operation
sender
receiver
first packet bit transmitted, t 0
last packet bit transmitted, t L / R
first packet bit arrives
RTT
last packet bit arrives, send ACK
ACK arrives, send next packet, t RTT L / R
38Pipelined 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
39Pipelining increased utilization
sender
receiver
first packet bit transmitted, t 0
last bit transmitted, t L / R
first packet bit arrives
RTT
last packet bit arrives, send ACK
last bit of 2nd packet arrives, send ACK
last bit of 3rd packet arrives, send ACK
ACK arrives, send next packet, t RTT L / R
Increase utilization by a factor of 3!
40Go-Back-N
- k-bit seq in pkt header
- window of up to N consecutive unacked pkts
allowed - send_base seq of the oldest unacked pkt
- nextseqnum seq of the next pkt to be sent
2k-1
0
41Go-Back-N
- Sender
- send pkt only when the window is not full
- ACK(n) ACKs all pkts up to, including seq n -
cumulative ACK - a single timer for the oldest transmitted but not
yet ACKed pkt - on timeout resends all pkts that have been sent
but havent been ACKed. - when ACK received
- restart timer if there are transmitted but not
yet ACKed pkts - stop the timer otherwise
42GBN sender extended FSM
rdt_send(data)
if (nextseqnum lt baseN) sndpktnextseqnum
make_pkt(nextseqnum,data,chksum)
udt_send(sndpktnextseqnum) if (base
nextseqnum) start_timer nextseqnum
else refuse_data(data)
L
base1 nextseqnum1
timeout
start_timer udt_send(sndpktbase) udt_send(sndpkt
base1) udt_send(sndpktnextseqnum-1)
rdt_rcv(rcvpkt) corrupt(rcvpkt)
L
rdt_rcv(rcvpkt) notcorrupt(rcvpkt)
base getacknum(rcvpkt)1 If (base
nextseqnum) stop_timer else start_timer
43GBN receiver extended FSM
default
udt_send(sndpkt)
rdt_rcv(rcvpkt) notcurrupt(rcvpkt)
hasseqnum(rcvpkt,expectedseqnum)
L
Wait
extract(rcvpkt,data) deliver_data(data) sndpkt
make_pkt(expectedseqnum,ACK,chksum) udt_send(sndpk
t) expectedseqnum
expectedseqnum1 sndpkt
make_pkt(0,ACK,chksum)
- ACK-only always send ACK for the most recently
received in-order pkt - may generate duplicate ACKs
- need only remember expectedseqnum, the seq of
the next in-order pkt - out-of-order pkt
- discard (dont buffer) -gt no receiver buffering
- Re-ACK pkt with highest in-order seq
44GBN inaction
Window size4
45Selective 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, limit seq s of sent,
unACKed pkts - receiver window
- N consecutive seq s permitted to accept
46Selective repeat sender, receiver windows
47Selective repeat
- pkt n in rcvbase, rcvbaseN-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
- data from above
- if next available seq in window, send pkt
- timeout(n)
- resend pkt n, restart timer
- ACK(n) in sendbase,sendbaseN-1
- mark pkt n as received
- if n is smallest unACKed pkt, advance window base
to next unACKed seq
48Selective repeat in action
49Selective repeat dilemma
- Example
- seq s 0, 1, 2, 3
- window size3
- receiver sees no difference in two scenarios!
- incorrectly passes duplicate data as new in (a)
- Q what relationship between seq space size and
window size?
50Summary of reliable data transfer mechanisms
- Checksum detect bit errors
- Timer detect packet loss, may cause duplicate
packets - Sequence number allow receiver to detect
duplicate packets, in-order packet delivery to
upper layer - Acknowledgement carry the seq of packet being
ACKed, individual/cumulative ACK - Window, pipelining improved sender utilization,
flow/congestion control
51Chapter 3 outline
- 3.1 Transport-layer services
- 3.2 Multiplexing and demultiplexing
- 3.3 Connectionless transport UDP
- 3.4 Principles of reliable data transfer
- 3.5 Connection-oriented transport TCP
- segment structure
- reliable data transfer
- flow control
- connection management
- 3.6 Principles of congestion control
- 3.7 TCP congestion control
52TCP Overview RFCs 793, 1122, 1323, 2018, 2581
- reliable, in-order byte steam
- no message boundaries
- pipelined
- TCP congestion and flow control set window size
- full duplex data transfer
- bi-directional data flow in same connection
- point-to-point
- one sender, one receiver
- connection-oriented
- handshaking (exchange of control msgs) initialize
sender, receiver state before data exchange - send receive buffers
- MSS maximum segment size
53TCP segment structure
URG urgent data (generally not used)
counting by bytes of data (not segments!)
ACK ACK valid
PSH push data now (generally not used)
bytes rcvr willing to accept
RST, SYN, FIN connection setup and teardown
RFC 793, 1323
Internet checksum (as in UDP)
54TCP seq. s and ACKs
- Seq. s
- byte stream number of first byte in segments
data - ACKs
- seq of next byte expected from other side
- cumulative ACK
- Q how receiver handles out-of-order segments
- A TCP spec doesnt say, - up to implementer
Host B (Server)
Host A (Client)
User types C
Seq42, ACK79, data C
host ACKs receipt of C, echoes back C
Seq79, ACK43, data C
host ACKs receipt of echoed C
Seq43, ACK80
simple telnet scenario
55TCP Round Trip Time and Timeout
- Q how to estimate RTT?
- SampleRTT measured time from segment
transmission until ACK receipt - ignore retransmissions
- SampleRTT will vary, want estimated RTT
smoother - average SampleRTT values
- Q how to set TCP timeout value?
- too short premature timeout
- unnecessary retransmissions
- too long slow reaction to segment loss
- longer than RTT
- but RTT varies
56TCP Round Trip Time and Timeout
EstimatedRTT (1- ?)EstimatedRTT ?SampleRTT
- Exponential weighted moving average
- influence of past sample decreases exponentially
fast - typical value ? 0.125
57Example RTT estimation
58TCP Round Trip Time and Timeout
- Setting the timeout
- EstimtedRTT plus safety margin
- large variation in EstimatedRTT -gt larger safety
margin - first estimate how much SampleRTT deviates from
EstimatedRTT
DevRTT (1-?)DevRTT
?SampleRTT-EstimatedRTT (typically, ? 0.25)
Then set timeout interval
TimeoutInterval EstimatedRTT 4DevRTT
59Chapter 3 outline
- 3.1 Transport-layer services
- 3.2 Multiplexing and demultiplexing
- 3.3 Connectionless transport UDP
- 3.4 Principles of reliable data transfer
- 3.5 Connection-oriented transport TCP
- segment structure
- reliable data transfer
- flow control
- connection management
- 3.6 Principles of congestion control
- 3.7 TCP congestion control
60TCP reliable data transfer
- TCP creates reliable data transfer service on top
of IPs unreliable service - Pipelined segments
- Cumulative acks
- Single retransmission timer
- Retransmissions are triggered by
- timeout events
- duplicate acks
- Initially consider simplified TCP sender
- ignore duplicate acks
- ignore flow control, congestion control
61TCP sender events
- data rcvd from app
- create segment with seq NextSeqNum
- start timer if not already running (think of
timer as for oldest unacked segment) - expiration interval TimeOutInterval
- timeout
- retransmit not-yet-acked segment with smallest
seq - restart timer
- Ack(y) rcvd
- If acknowledges previously unacked segments (y gt
SendBase) - Set SendBase y
- start timer if there are outstanding segments
62TCP sender(simplified)
NextSeqNum InitialSeqNum
SendBase InitialSeqNum loop (forever)
switch(event) event
data received from application above
create TCP segment with sequence number
NextSeqNum if (timer currently
not running) start timer
pass segment to IP
NextSeqNum NextSeqNum length(data)
event timer timeout
retransmit not-yet-acknowledged segment with
smallest sequence number
start timer event ACK
received, with ACK field value of y
if (y gt SendBase)
SendBase y if (there are
currently not-yet-acknowledged segments)
start timer
/ end of loop forever /
- Comment
- SendBase-1 last
- cumulatively acked byte
- Example
- SendBase 71y 73, so the rcvrwants 73 y
gt SendBase?new - data (71,72) is acked
63TCP retransmission scenarios
Host A
Host B
Seq92, 8 bytes data
Seq100, 20 bytes data
ACK100
ACK120
Seq92, 8 bytes data
Sendbase 100
SendBase 120
discard segment
ACK120
Seq92 timeout
SendBase 100
SendBase 120
premature timeout
64TCP retransmission scenarios (more)
SendBase 120
65TCP ACK generation RFC 1122, RFC 2581
TCP Receiver action Delayed ACK. Wait up to
500ms for another in-order segment. If no such
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
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
66Fast 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 a segment is lost, there will likely be many
duplicate ACKs.
- If sender receives 3 duplicate ACKs for the same
data, it supposes that segment after ACKed data
was lost - fast retransmit resend segment before timer
expires
67Fast retransmit algorithm
event ACK received, with ACK field value of y
if (y gt SendBase)
SendBase y
if (there are currently not-yet-acknowledged
segments) start
timer
else increment count
of dup ACKs received for y
if (count of dup ACKs received for y 3)
resend segment with
sequence number y
a duplicate ACK for already ACKed segment
fast retransmit
68Chapter 3 outline
- 3.1 Transport-layer services
- 3.2 Multiplexing and demultiplexing
- 3.3 Connectionless transport UDP
- 3.4 Principles of reliable data transfer
- 3.5 Connection-oriented transport TCP
- segment structure
- reliable data transfer
- flow control
- connection management
- 3.6 Principles of congestion control
- 3.7 TCP congestion control
69TCP Flow Control
- receive side of TCP connection has a receive
buffer
- speed-matching service matching the send rate to
the receiving apps drain rate
- app process may be slow at reading from buffer
70TCP Flow control how it works
- Rcvr advertises spare room by including value of
RcvWindow in segments - Sender limits unACKed data to RcvWindow
- LastByteSent-LastByteAcked lt RcvWindow
guarantees receive buffer doesnt overflow
- (Suppose TCP receiver discards out-of-order
segments) - spare room in buffer
- RcvWindow
- RcvBuffer-LastByteRcvd - LastByteRead
71Chapter 3 outline
- 3.1 Transport-layer services
- 3.2 Multiplexing and demultiplexing
- 3.3 Connectionless transport UDP
- 3.4 Principles of reliable data transfer
- 3.5 Connection-oriented transport TCP
- segment structure
- reliable data transfer
- flow control
- connection management
- 3.6 Principles of congestion control
- 3.7 TCP congestion control
72TCP Connection Management
- Three way handshake
- Step 1 client host sends TCP SYN segment to
server - specifies initial seq
- no data
- Step 2 server host receives SYN, replies with
SYNACK segment - server allocates buffers, variables
- specifies server initial seq.
- Step 3 client receives SYNACK, allocate buffers
and variables, replies with ACK segment, which
may contain data
- 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
- server contacted by client
73TCP Connection Management Three way handshake
client
server
Connection request
SYN1, seqclient_ins
Connection granted
SYN1, seqserver_ins, Ackclient_ins1
ACK
SYN0, seqclient_ins1, Ackserver_ins1
74TCP Connection Management (cont.)
- Closing a connection
- client closes socket
- Step 1 client end system sends TCP FIN control
segment to server, FIN bit set to 1 - Step 2 server receives FIN, replies with ACK.
Closes connection, sends FIN.
client
server
closing
FIN
ACK
closing
FIN
ACK
timed wait
closed
closed
75TCP Connection Management (cont.)
- Step 3 client receives FIN, replies with ACK.
- Enters timed wait resend the final ACK in case
it is lost - Connection closes after wait
- Step 4 server receives ACK. Connection closed.
client
server
closing
FIN
ACK
closing
FIN
ACK
timed wait
closed
closed
76TCP Connection Management (cont)
TCP server lifecycle
TCP client lifecycle
77Chapter 3 outline
- 3.1 Transport-layer services
- 3.2 Multiplexing and demultiplexing
- 3.3 Connectionless transport UDP
- 3.4 Principles of reliable data transfer
- 3.5 Connection-oriented transport TCP
- segment structure
- reliable data transfer
- flow control
- connection management
- 3.6 Principles of congestion control
- 3.7 TCP congestion control
78Principles 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)
79Causes/costs of congestion scenario 1
Host C
- two senders, two receivers
- one router, link capacity C, infinite buffers
- no retransmission
Host D
- large delays when packet arrival rate nears link
capacity
80Causes/costs of congestion scenario 2
- one router, finite buffers
- sender retransmits lost packets
Host A
lout
lin original data
Host C
l'in original data, plus retransmitted data
Host B
finite shared output link buffers
Host D
81Causes/costs of congestion scenario 2
- Case 1 send a packet only when a buffer is free
- Case 2 perfect retransmission only when loss
- Case 3 retransmission of delayed (not lost)
packet makes larger (than case 2) for
same
C/2
C/3
C/4
R/2
R/2
R/2
- costs of congestion
- more work performed by sender (retransmissions)
- unneeded retransmissions (due to large delay)
82Causes/costs of congestion scenario 3
- four senders
- multihop paths
- timeout/retransmit
Q what happens as and increase ?
lin original data
Host B
l'in original data, plus retransmitted data
finite shared output link buffers
lout
Host C
83Causes/costs of congestion scenario 3
- Another cost of congestion
- when packet dropped, any upstream transmission
capacity used for that packet was wasted!
84Approaches towards congestion control
Two broad approaches towards congestion control
- network-assisted congestion control
- routers provide feedback to end systems
- single bit indicating congestion (SNA, DECnet,
TCP/IP ECN, ATM) - explicit rate sender should send at (ATM)
- end-end congestion control
- no explicit feedback from network
- congestion inferred from end-system observed
loss, delay - approach taken by TCP
85Case study ATM ABR congestion control
- ABR available bit rate
- elastic service
- if senders path under-loaded
- sender should use available bandwidth
- if senders path congested
- sender throttled to minimum transmission 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 (severe congestion)
- RM cells returned to sender by receiver
-
86Case study ATM ABR congestion control
- two-byte ER (explicit rate) field in RM cell
- congested switch may lower ER value in cell
- EFCI bit in data cells set to 1 in congested
switch - if data cell preceding RM cell has EFCI set,
destination sets CI bit in returned RM cell - Source computes its sending rate as a function of
CI, NI, and ER values in a returned RM cell
87Chapter 3 outline
- 3.1 Transport-layer services
- 3.2 Multiplexing and demultiplexing
- 3.3 Connectionless transport UDP
- 3.4 Principles of reliable data transfer
- 3.5 Connection-oriented transport TCP
- segment structure
- reliable data transfer
- flow control
- connection management
- 3.6 Principles of congestion control
- 3.7 TCP congestion control
88TCP Congestion Control
- end-end control (no network assistance)
- sender limits transmission rate
- LastByteSent-LastByteAcked
- ? minCongWin, RcvWindow)
- Roughly,
- CongWin is dynamic, function of perceived network
congestion
- How does sender perceive congestion?
- loss event timeout or 3 duplicate acks
- TCP sender reduces rate (CongWin) after loss
event - three mechanisms
- AIMD
- slow start
- reaction to timeout events
89TCP AIMD
- additive increase
- increase CongWin by 1 MSS every RTT in the
absence of loss events probing - also called congestion avoidance
- multiplicative decrease cut CongWin in half
after loss event
Long-lived TCP connection
90TCP Slow Start
- When connection begins, increase rate
exponentially fast until first loss event
- When connection begins, CongWin 1 MSS
- Example MSS 500 bytes RTT 200 msec
- initial rate 20 kbps
- available bandwidth may be gtgt MSS/RTT
- desirable to quickly ramp up to respectable rate
91TCP Slow Start (more)
- When connection begins, increase rate
exponentially until first loss event - double CongWin every RTT
- done by incrementing CongWin by one MSS for every
ACK received - Summary initial rate is slow but ramps up
exponentially fast
92Refinement
Philosophy
- After 3 dup ACKs
- CongWin is cut in half
- window then grows linearly
- But after timeout event
- CongWin instead set to 1 MSS
- window then grows exponentially
- to a threshold, then grows linearly
- Implementation At loss event, threshold is set
to 1/2 of CongWin just before loss event
- 3 dup ACKs indicates network capable of
delivering some segments - timeout before 3 dup ACKs is more alarming
93Summary TCP Congestion Control
- When CongWin is below Threshold, sender in
slow-start phase, window grows exponentially. - When CongWin is above Threshold, sender is in
congestion-avoidance phase, window grows
linearly. - When a triple duplicate ACK occurs, Threshold set
to CongWin/2 and CongWin set to Threshold. - When timeout occurs, Threshold set to CongWin/2
and CongWin is set to 1 MSS.
Triple duplicate ACK
94Chapter 3 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 layers) - into the network core