1. Modified by John Copeland
Chapter 3 Transport Layer
Modified by John Copeland
Georgia Tech
for use in ECE3600
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All material copyright
J.F Kurose and K.W. Ross, All Rights Reserved
Computer Networking: A Top Down Approach Featuring the Internet, 5th edition. Jim Kurose, Keith Ross Addison-Wesley, July 2009.
Transport Layer

2. Chapter 3: Transport Layer
Our goals:
understand principles behind transport layer services:
multiplexing/demulti-plexing
reliable data transfer
flow control
congestion control
learn about transport layer protocols in the Internet:
UDP: connectionless transport
TCP: connection-oriented transport
TCP congestion control
Transport Layer

3. Chapter 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
Transport Layer

4. Transport services and protocols
provide logical communication between app processes running on different hosts
transport protocols run in end systems
send side: breaks app messages into segments, passes to network layer
receive side: reassembles segments into messages, passes to app layer
more than one transport protocol available to apps
Internet: TCP and UDP
application
transport
network
data link
physical
network
data link
physical
network
data link
physical
network
data link
physical
logical end-end transport
network
data link
physical
network
data link
physical
application
transport
network
data link
physical
Transport Layer

5. Transport vs. network layer
Household analogy:
12 kids sending letters to 12 kids
processes = kids
app messages = letters in envelopes
hosts = houses
transport protocol = Ann and Bill
network-layer protocol = postal service
network layer: logical communication between hosts
transport layer: logical communication between processes*
relies on, enhances, network layer services
* running on different hosts
Transport Layer

6. Internet transport-layer protocols
reliable, in-order delivery (TCP)
congestion control
flow control
connection setup
unreliable, unordered delivery: UDP
no-frills extension of “best-effort” IP
services not available:
delay guarantees
bandwidth guarantees
application
transport
network
data link
physical
network
data link
physical
network
data link
physical
network
data link
physical
logical end-end transport
network
data link
physical
network
data link
physical
application
transport
network
data link
physical
Transport Layer

7. Chapter 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
Transport Layer

8. Multiplexing/demultiplexing
Multiplexing at send host:
Demultiplexing at rcv host:
delivering received segments
to correct socket
gathering data from multiple
sockets, enveloping data with
header (later used for
demultiplexing)
= socket
= process
application
application
application P3 P1 P4 P1 P2
transport
transport
transport
network
network
network
link
link
link
physical
physical
physical
host 3
host 1
host 2
Transport Layer

9. How demultiplexing works
host receives IP datagrams
each datagram has source IP address, destination IP address
each datagram carries 1 transport-layer segment
each segment has source, destination port number
host uses IP addresses & port numbers to direct segment to appropriate socket
32 bits
source port #
dest port #
other header fields
application
data
(message)
TCP and UDP segment format
Transport Layer

10. Connectionless demultiplexing
Create sockets with port numbers:
DatagramSocket mySocket1 = new DatagramSocket(12534);
DatagramSocket mySocket2 = new DatagramSocket(12535);
UDP socket identified by two-tuple
(plus "UDP", it is a 3-tuple):
(For incoming segment:
destination IP address,
destination 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 UDP socket
This means a single process thread can handle all UDP packets coming to a host that have the same local (destination) UDP port number.
This is not the case for TCP (an addition thread is needed for each remote IP address and TCP port pair).
Transport Layer

11. Connectionless (UDP) demux (cont)
DatagramSocket serverSocket = new DatagramSocket(6428);
Client
IP:B P2
client
IP: A P1 P3
server
IP: C
SP: 6428
DP: 9157
SP: 9157
DP: 6428
DP: 5775
SP: 5775
Same incoming UDP destination port,
delivered to same process, the one
that opened that socket.
SP provides “return address”
Transport Layer

12. Connection-oriented demux
TCP socket identified by a 4-tuple
(+ "TCP" -> a 5-tuple):
source IP address
source port number
dest IP address
dest port number
recv host TCP layer uses all four values to direct segment to appropriate socket
Server host may support many simultaneous TCP sockets:
each TCP socket identified by its own 4-tuple (5-tuple)
Web servers have different sockets for each connecting client
non-persistent HTTP will have different socket (different client port number) for each request
Transport Layer

13. Connection-oriented (TCP) demux (cont)
Different TCP source port
from P5 and P6 on client C
-> different process (thread)
(P2, P3 on server B) P1
server
IP: A P4 P5 P6 P2 P1 P3
SP: 5775
DP: 80
S-IP: C
D-IP:B
SP:5776
SP: 5777
DP: 80
DP: 80
server
IP:B
client
IP: C
S-IP: C
S-IP: C
D-IP:A
D-IP:B
Transport Layer

14. Connection-oriented (TCP) demux: Threaded Web Server
Different TCP source port
-> different sub-process (thread)
Listening
Socket P1
client
IP: A P4 P2 P1 P3
SP: *
DP: 80 *
SP: 5775
DP: 80
S-IP: *
S-IP: B
D-IP:C
* = "any"
D-IP:C
SP: 9157
SP: 9157
DP: 80
DP: 80
Client
IP:B
server
IP: C
S-IP: A
S-IP: B
D-IP:C
D-IP:C
thread, spawned or "forked" from main process,
manages each connected socket.
Transport Layer

15. Active Socket states: ESTABLISHED or LISTEN (netstat)
mc:/Users/copeland root# netstat -f inet -al
Active Internet connections
Proto Recv-Q Send-Q Local Address (IP.port) Foreign Address (state)
tcp mc ESTABLISHED
tcp mc.ssh home ESTABLISHED
tcp * *.* LISTEN
tcp localhost.ipp *.* LISTEN
tcp *.afpovertcp *.* LISTEN
tcp *.timbuktu *.* LISTEN
tcp *.ssh *.* LISTEN
udp *.mdns *.*
udp * *.*
udp * *.*
udp *.ipp *.*
udp localhost localhost.1022
udp localhost localhost.1022
udp localhost *.*
udp localhost *.*
udp *.timbuktu-srv *.*
udp *.timbuktu *.*
udp *.* *.*
udp mc.ntp *.*
udp localhost.ntp *.*
udp *.ntp *.*
udp *.tftp *.*
Well Known Ports (from /etc/services): ssh = 22, mdns = 5353, timbuktu = 407, timbuktu-srv3 = 1419, ntp = 123, tftp = 69, ipp = 631
Host names (from /etc/hosts): mc, home (from DNS)
Transport Layer

16. Chapter 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
Transport Layer

17. 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
Transport Layer

18. UDP: more other UDP uses often used for streaming multimedia apps
loss tolerant
rate sensitive
other UDP uses
DNS (name lookup)
SNMP (network management)
NTP (network time protocol)
to get 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
length
checksum
Application
data
(message)
UDP segment format
Transport Layer

19. UDP checksum
Goal: detect “errors” (e.g., flipped bits) in transmitted segment
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 nonetheless? More later ….
Transport Layer

20. Internet Checksum Example
Note
When adding numbers, a carryout from the most significant bit needs to be added to the result
Example: add two 16-bit integers (1's compliment)
Kurose and Ross forgot to say anything about wrapping the carry and adding it to low order bit
wraparound
sum
checksum
Transport Layer

21. Chapter 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
Transport Layer

22. Principles of Reliable data transfer
important in app., transport, link layers
top-10 list of important networking topics!
characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)
Transport Layer

23. Principles of Reliable data transfer
important in app., transport, link layers
top-10 list of important networking topics!
characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)
Transport Layer

24. Principles of Reliable data transfer
important in app., transport, link layers
top-10 list of important networking topics!
characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)
Transport Layer

25. Reliable data transfer: getting started
rdt_send(): called from above, (e.g., by app.). Passed data to
deliver to receiver upper layer
deliver_data(): called by rdt to deliver data to upper
send
side
receive
side
udt_send(): called by rdt,
to transfer packet over
unreliable channel to receiver
rdt_rcv(): called when packet arrives on rcv-side of channel
Transport Layer

26. Reliable data transfer: getting started
We’ll:
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
state 1
state: when in this “state” next state uniquely determined by next event
state 2
event
actions
Transport Layer

27. Rdt1.0: 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 read data from underlying channel
Event causes change of state
Action that occurs
Wait for call from above
rdt_send(data)
Wait for call from below
rdt_rcv(packet)
extract (packet,data)
deliver_data(data)
packet = make_pkt(data)
udt_send(packet)
sender
receiver
Transport Layer

28. Rdt2.0: channel with bit errors
underlying channel may flip bits in packet
checksum to detect bit errors
the question: how to recover from errors:
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->sender
Transport Layer

29. rdt2.0: FSM specification
rdt_send(data)
snkpkt = make_pkt(data, checksum)
udt_send(sndpkt)
receiver
rdt_rcv(rcvpkt) &&
isNAK(rcvpkt)
Wait for ACK or NAK
Wait for call from above
udt_send(NAK)
rdt_rcv(rcvpkt) &&
corrupt(rcvpkt)
udt_send(sndpkt)
rdt_rcv(rcvpkt) && isACK(rcvpkt)
Wait for call from below L
sender
rdt_rcv(rcvpkt) &&
notcorrupt(rcvpkt)
extract(rcvpkt,data)
deliver_data(data)
udt_send(ACK)
Transport Layer

30. rdt2.0: operation with no errors
rdt_send(data)
snkpkt = make_pkt(data, checksum)
udt_send(sndpkt)
rdt_rcv(rcvpkt) &&
isNAK(rcvpkt)
Wait for ACK or NAK
Wait for call from above
udt_send(NAK)
rdt_rcv(rcvpkt) &&
corrupt(rcvpkt)
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(ACK)
Transport Layer

31. rdt2.0: error scenario rdt_send(data)
snkpkt = make_pkt(data, checksum)
udt_send(sndpkt)
rdt_rcv(rcvpkt) &&
isNAK(rcvpkt)
Wait for ACK or NAK
Wait for call from above
udt_send(NAK)
rdt_rcv(rcvpkt) &&
corrupt(rcvpkt)
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(ACK)
Transport Layer

32. rdt2.0 has a fatal flaw! What happens if ACK/NAK corrupted?
sender doesn’t know what happened at receiver!
can’t just retransmit: possible duplicate
Handling duplicates:
sender retransmits current pkt if ACK/NAK garbled
sender adds sequence number to each pkt
receiver discards (doesn’t deliver up) duplicate pkt
stop and wait
Sender sends one packet,
then waits for receiver
response
Transport Layer

33. rdt2.1: sender, handles garbled ACK/NAKs
rdt_send(data)
sndpkt = make_pkt(0, data, checksum)
udt_send(sndpkt)
rdt_rcv(rcvpkt) &&
( corrupt(rcvpkt) ||
isNAK(rcvpkt) )
Wait for ACK or NAK 0
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
Wait for ACK or NAK 1
Wait for
call 1 from above
rdt_rcv(rcvpkt) &&
( corrupt(rcvpkt) ||
isNAK(rcvpkt) )
rdt_send(data)
sndpkt = make_pkt(1, data, checksum)
udt_send(sndpkt)
udt_send(sndpkt)
Transport Layer

34. rdt2.1: 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)
Wait for
0 from below
Wait for
1 from below
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)
Transport Layer

35. rdt2.1: discussion Sender: seq # added to pkt
two seq. #’s (0,1) will suffice. Why?
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 #
note: receiver can not know if its last ACK/NAK received OK at sender
Transport Layer

36. rdt2.2: 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 [say “0”]
duplicate ACK at sender [two ACKs for “0”] results in same action as NAK: retransmit current pkt [“1”]
Transport Layer

37. rdt2.2: sender, receiver fragments
rdt_send(data)
sndpkt = make_pkt(0, data, checksum)
udt_send(sndpkt)
rdt_rcv(rcvpkt) &&
( corrupt(rcvpkt) ||
isACK(rcvpkt,1) )
Wait for call 0 from above
Wait for ACK
udt_send(sndpkt)
sender FSM
fragment
rdt_rcv(rcvpkt)
&& notcorrupt(rcvpkt)
&& isACK(rcvpkt,0)
rdt_rcv(rcvpkt) &&
(corrupt(rcvpkt) ||
has_seq1(rcvpkt)) L
Wait for
0 from below
receiver FSM
fragment
sndpkt =
make_pkt(ACK0, chksum)
udt_send(sndpkt)
“Duplicate ACK”
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt)
&& has_seq1(rcvpkt)
extract(rcvpkt,data)
deliver_data(data)
sndpkt = make_pkt(ACK1, chksum)
udt_send(sndpkt)
Transport Layer

38. rdt3.0: channels with errors and loss
New assumption: underlying channel can also lose packets (data or ACKs)
checksum, seq. #, ACKs, retransmissions will be of help, but not enough
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
receiver must specify seq # of pkt being ACKed
requires countdown timer
Transport Layer

39. rdt3.0 sender L L L L rdt_send(data) rdt_rcv(rcvpkt) &&
( corrupt(rcvpkt) ||
isACK(rcvpkt,1) )
sndpkt = make_pkt(0, data, checksum)
udt_send(sndpkt)
start_timer
rdt_rcv(rcvpkt) L L
Wait for
call 0 from above
Wait for ACK0
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
Wait for ACK1
Wait for
call 1 from above
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(sndpkt)
start_timer L
Transport Layer

40. rdt3.0 in action
Transport Layer

41. rdt3.0 in action
Transport Layer

42. Review rdt3.0 ACK Packet Lets Sender know that data has been received.
Sequence Number
In Header - Lets Receiver know where to place data in buffer
(to avoid duplication and gaps)
Acknowledgement Number
In ACK - Lets Sender know which data has been received.
(so it knows what to resend)
Retransmit Timer
Keeps connection from freezing because of a lost packet.
Transport Layer

43. Performance 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 = =
= millisec
transmit
R (transmission rate, bps)
10**9 b/sec
U sender: utilization – fraction of time sender busy sending
1KB pkt every 30 msec -> 33kB/sec thruput over 1 Gbps link
network protocol limits use of physical resources!
Transport Layer

44. rdt3.0: 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
U is the “channel utilization factor”, or “efficiency”
Transport Layer

45. 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
Transport Layer

46. Pipelining: 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 1
last bit of 2nd packet arrives, send ACK 2
last bit of 3rd packet arrives, send ACK 3
ACK arrives, send next
packet, t = RTT + L / R
Increase utilization
by a factor of 3 !
If 3L/R < RTT and Window = 3L, then Throughput Rt = R*U = (Window/RTT)
Transport Layer

47. Go-Back-N Sender: 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 receive duplicate ACKs (see receiver)
timer for each in-flight pkt
timeout(n): retransmit pkt n and all higher seq # pkts in window
Transport Layer

48. Go Back N: Receiver
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 packet with the highest in-order seq #
Transport Layer

49. GBN in action
Transport Layer

50. 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
Transport Layer

51. Selective repeat: sender, receiver windows
send_base moves forward to first not-ack’ed packet
(TCP buffers all, but ACK’s first missing byte, = rcv-base)
rcv_base moves forward to first not-transferred packet
Transport Layer

52. Selective repeat receiver 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]:
mark pkt n as received
if n smallest unACKed pkt, advance window base to next unACKed seq #
pkt 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) (already received & delivered)
otherwise:
Ignore
Transport Layer

53. Selective repeat in action
TCP
ack1
ack2
ACK(2nd)
ack2 (3rd)
ack2 (4th)
On 4th ack2
The sender resends only
Pkt2,
Then continues
Sender must keep a list
of pkt’s ACKed
Transport Layer

54. 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?
Transport Layer

55. Reliable Data Transport
Problem
Packet may arrive with errors.
Packet may not arrive.
Sender may wait forever for ACK.
ACK may not arrive, dup. sent.
Packets may arrive out-of-order.
Inefficient to send one pkt per RT
Missing packet early in window.
“Go-Back-N” inefficient.
---- Also in TCP ---
Packets may be different sizes.
Slow down when network congested (as detected by RTO or triple duplicate ACKs.
Know when receiver buffer will be full.
Solution
Add checksum, CRC, or hash.
Receiver sends “ACK” back. If
ACK not received, packet re-sent.
Timeout timer added to sender.
Add sequence no.s to detect dups.
Buffer packets to rearrange order.
Have a “window” to send before ACK
(pipelining).
“Go-Back-N” to last in-order packet.
“Selective Repeat” to fill in gaps only.
----
Sequence number for each byte.
“Slow-Start”, or "Multiplicative Decrease" to reduce transmit window.
Receiver includes “space left” in every ACK.
Transport Layer
3-55

56. Chapter 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
Transport Layer

57. TCP: Overview RFCs: 793, 1122, 1323, 2018, 2581 point-to-point:
one sender, one receiver
reliable, in-order byte steam:
no “message boundaries”
pipelined:
TCP congestion and flow control set window size
send & receive buffers
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
Transport Layer

58. TCP segment structure source port # dest port # sequence number
32 bits
application
data
(variable length)
sequence number
acknowledgement number
Receive window
Urg data pnter
checksum F S R P A U
head
len
not
used
Options (variable length)
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 estab
(setup, teardown
commands)
Seg. Size = IP Size
– IP Header Size
Internet
checksum
(as in UDP)
Data Size = IP Size
– IP Header Size
- TCP Header Size
Transport Layer

59. TCP seq. #’s and ACKs Seq. #’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 the implementer
Host A
Host B
User
types
‘C’
Seq=42, ACK=79, data = ‘Hi’
host ACKs
receipt of
‘C’, echoes
back ‘C’
Seq=79, ACK=44, data =‘Hello’
host ACKs
receipt
of echoed
‘C’
Seq=44, ACK=84
time
simple telnet scenario
Transport Layer

60. TCP Round Trip Time and Timeout
Q: how to set TCP timeout value?
longer than RTT
but RTT varies
if too short: premature timeout and
unnecessary retransmissions
if too long: slow reaction to segment loss
Q: how to estimate RTT?
SampleRTT: measured time from segment transmission until ACK receipt
ignore measurement if packet was retransmitted.
SampleRTT will vary, we want the estimated RTT “smoother”
average several recent measurements, not just current SampleRTT by using a "running average.
Transport Layer

61. TCP Round Trip Time and Timeout
EstimatedRTT = (1- )*EstimatedRTT + *SampleRTT
This is like a computer instruction, rather than an algebra. It may be clearer to write:
EstimatedRTT[new] = (1-)* EstimatedRTT[old]
+ *SampleRTT[new]
where "old" and "new" are array indices, and
new = old + 1
Exponential weighted moving average
influence of past sample decreases exponentially fast
typical value:  = 0.125
Transport Layer

62. Example RTT estimation:
Transport Layer

63. TCP Round Trip Time and Timeout
Setting the retransmission timeout, RTO:
use EstimatedRTT plus “safety margin”
larger variation in EstimatedRTT -> larger safety margin
use a running average, DevRTT, to estimate how much SampleRTT deviates from EstimatedRTT:
DevRTT[new] = (1-)* DevRTT[old] +
 * |SampleRTT[new]-EstimatedRTT[old*]|
(typically,  = 0.25) Note the absolute bars,|...|
Then set a new timeout interval, RTO:
Timeout Interval: RTO = EstimatedRTT + 4 * DevRTT
*Note - "old" value used, not "new" value.
Transport Layer

64. A[i] is a exponentially-decaying (1-c) weighed
average of x[i], x[i-1], x[i-2], …
A[i] = (1-c) * A[ i – 1 ] + c * x[ i ]
A[ i - 1] = (1-c) * A[ i – 2 ] + c * x[ i - 1 ]
A[ i - 2] = (1-c) * A[ i – 3 ] + c * x[ i - 2 ]
. . .
A[i] = c * { x[ i ] + (1-c) * x[ i – 1 ] + (1-c)^2 * x[ i – 2 ] + … }
Advantages: Only the last value of A[i] needs to be remembered.
More recent values of x[i] are more heavily weighted. 64

65. Chapter 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
Transport Layer

66. TCP reliable data transfer
TCP creates rdt service on top of IP’s unreliable service
Pipelined segments
Window is in bytes
Cumulative acks*
TCP uses single retransmission timer*
Retransmissions are triggered by:
timeout events (Go-Back-N)
3 duplicate acks (like Selective Repeat*)
Differs from Selective Repeat. If a segment is missing, the receiver keeps sending the same ACK number (the first missing byte).
After 3 duplicate ACKs, the sender sends only the missing segment.
Transport Layer

67. Maximum Segment Size (MSS), in bytes
The initial segments (the SYN and SYN-ACK) contain the MSS in an option field. It stays constant after this.
This tells the other host the maximum size of a segment that can be handled by their local network (without fragmentation).
Examples, one host may say it's MSS value is 1400, the other may say it's MSS value is 1420.
Since segments have to transverse both local networks, the smaller MSS value is used for the connection.
TCP rules, involving Window sizes, are in number of MSS (bytes), not number of segments.
For simplification, examples may say "the host is sending maximum size segments," so that 1 MSS = 1 segment. Sometimes this is implied without being stated in problems.
MSS includes the TCP header bytes (20 to 64) and data bytes, but not the IP header bytes (20). Since Ethernet and WiFi limit datagram size to 1500 bytes, MSS is never larger than 1480 bytes when either host is on a LAN.
3-67

68. 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
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 [don’t ACK every segment]
Immediately send single cumulative
ACK, ACKing both in-order segments
[ACK every other segment]
Immediately send duplicate ACK,
indicating seq. # of next expected byte
[1st byte in gap]
Immediate send ACK for highest
continuous byte. If segment does not
start at lower end of gap, it will be
another duplicate ACK.
Transport Layer

69. Two windows, separate limits on bytes sent (Flow Control & Congestion Control).
RcvrWin in each TCP header received (room in receiver buffer)
CongWin calculated by sender (in response to perceived congestion).
Sender Buffer
SendBase is the lowest unacknowledged byte no.,
and it is also the highest ACK no. received. 92
100
120
ACK 92,
Window 40
received
+ CongWin (30)
+ RcvrWin (40)
SendBase
= 92
+ CongWin (30)
ACK 120,
Window 20
received
+ RcvrWin (20)
SendBase
= 120
140
160
Sender can not send bytes beyond either window.
MSS limits bytes in a segment, constant after SYN / SYN-ACK.
Transport Layer
3-69

70. TCP Flow control LastByte LastByte InBuffer ACKed Receiver-Window =
Rcvr advertises spare room by including value of RcvWindow in every segment (TCP header)
Sender limits data to RcvWindow + ACK
guarantees receive buffer doesn’t overflow
LastByte
InBuffer
LastByte
ACKed
<- byte no.
higher byte no.s
Receiver-Window =
spare room in buffer
= LastByteInBuffer - LastByteACKed
sender won’t overflow
receiver’s buffer by
transmitting too fast
flow control
Transport Layer

71. TCP sender events: RTO (timeout)
data rcvd from app:
Create segment with seq #
seq # is byte-stream number of first data byte in segment
start RTO timer if not already running (think of timer as being for oldest un-ACKed segment)
expiration interval: calculate new value for TimeOutInterval(RTO)
timeout:
retransmit segment that caused timeout (Go-Back-N)
RTO -> 2*RTO temporarily
CongWin = 1 MSS (maximum segment size, e.g bytes)*
restart RTO timer
ACK rcvd:
If acknowledges previously un-ACKed segments
update what is known to be ACKed (move SendBase to ACK#)
CongWin =CongWin + MSS
RTO -> original value (from 2X)
start RTO timer if there are outstanding segments
* CongWin has become very small. It can double in size every RTT, but it would take 6 RTT to get back to 64 MSS (Slow Start mode).
Transport Layer

72. TCP: retransmission scenarios
Host A
Seq=92, 8 bytes data
ACK=100
loss
timeout
lost ACK scenario
Host B X
time
Host A
Host B
Seq=92 timeout
Seq=92, 8 bytes data
Seq=100, 20 bytes data
ACK=100
ACK=120
SendBase
= 100
Seq=92, 8 bytes data
SendBase
= 120
Seq=120, 10 bytes data
Seq=92 timeout
ACK=120
SendBase
= 100
SendBase
= 120
ACK=130
time
premature timeout
Transport Layer

73. TCP retransmission scenarios (more)
Host A
Seq=92, 8 bytes data
ACK=100
loss
timeout
"Cumulative ACK" scenario
Host B X
Seq=100, 20 bytes data
ACK=120
time
Sender Buffer 92
100
120
+ CongWin
+ RcvrWin
SendBase
SendBase
= 120
+ CongWin
+ RcvrWin’
SendBase
Sender can not send bytes
beyond either window.
Every ACK has updated value of RcvrWin
Transport Layer

74. Fast Retransmit
If sender receives 4 ACKs for the same data (3 dups), it supposes that segment after ACKed data was lost:
fast retransmit: resend segment before timer expires
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.
When resent packet is ACKed before a timeout, go to Fast Recovery Mode:
- Halve "CongWin"
- Increase CongWin by 1 MSS per CongWin bytes sent and ACKed.
Transport Layer

75. Fast retransmit algorithm:
event: ACK received, with ACK field value of y
if (y > 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
then resume sending new segments
CongWin = CongWin/2
a duplicate ACK for
already ACKed segment
fast retransmit
Transport Layer

76. Fast Retransmit Sequence Number (kBytes) Send time (ms)
(after 3 duplicate ACKs)
(not CongWin limited)
Sequence
Number
(kBytes)
Retransmission
Retransmission
Retransmission.
Send time (ms)

77. Chapter 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
Transport Layer

78. TCP Flow Control flow control
sender won’t overflow
receiver’s buffer by
transmitting too fast
flow control
receive side of TCP connection has a receive buffer:
<- byte no.
speed-matching service: matching the send rate to the receiving app’s drain rate
higher byte no.s
Every TCP header sent by the receiver contains the value of the receiver's Window. This decreases as the incoming data buffer fills up.
app process may be slow at reading from buffer
Transport Layer

79. Chapter 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
Transport Layer

80. Chapter 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
Transport Layer

81. Principles of Congestion Control
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)
Sender's CongWin decreases , throughput = CongWin/RTT decreases
long delays (queueing in router buffers)
RTT increases, throughput = Window/RTT decreases
Transport Layer

82. Causes/costs of congestion: scenario 1
unlimited shared output link buffers
Host A
lin : original data
Host B
lout
two senders, two receivers
one router, infinite buffers
no retransmission
large delays when congested
maximum achievable throughput
Transport Layer

83. Causes/costs of congestion: scenario 3
Host A
lout
Host B
Another “cost” of congestion:
when packet dropped, any “upstream transmission capacity used for that packet (to the point where it was dropped) was wasted!
Transport Layer

84. Approaches towards congestion control
Two broad approaches towards congestion control:
End-end congestion control:
no explicit feedback from network
congestion inferred from end-system (1) observed loss, and (2) delay – by two methods: (a) duplicate ACKs or (b) RTOs.
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
Transport Layer

85. 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
Transport Layer

86. Chapter 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
Transport Layer

87. TCP congestion control: additive increase, multiplicative decrease
Approach: increase transmission rate (window size), probing for usable bandwidth, until loss occurs
additive increase: increase CongWin by 1 MSS every RTT until loss detected
multiplicative decrease: cut CongWin in half after loss
Lost packet (3 duplicate ACKs)
Saw tooth
behavior: probing
for bandwidth
congestion window size
time
* MSS = maximum segment size (e.g.. , 1280 bytes). specified as a TCP option in SYN and SYN-ACK packets.
Transport Layer

88. TCP Congestion Control: details
sender limits transmission:
LastByteSent-LastByteAcked
 CongWin
Roughly,
CongWin is dynamic, function of perceived network congestion
How does sender perceive congestion?
loss event = timeout (RTO) or 3 duplicate ACKs
TCP sender reduces rate (CongWin) after loss event
mechanisms:
AIMD (additive increase, multiplicative decrease)
slow start (after RTO)
conservative after timeout events (CongWin = 1 MSS)
rate =
CongWin
RTT
Bytes/sec
Transport Layer

89. TCP Slow Start When connection begins, CongWin = 1 MSS*
Example: MSS = 500 bytes & RTT = 200 msec
initial rate = 20 kbps
available bandwidth may be >> MSS/RTT
desirable to quickly ramp up to respectable rate
When connection begins, increase rate exponentially fast until first loss event (doubles every RTT)
or until CongWin = Receiver-Window.
* MSS = maximum segment size (e.g.. , 1420 bytes). specified as a TCP option in SYN and SYN-ACK packets.
Transport Layer

90. TCP Slow Start (more)
When connection begins, increase rate exponentially until first loss event:
double CongWin every RTT
done by incrementing CongWin for every ACK received (by d-SendBase)
Summary: initial rate is slow but ramps up exponentially fast
Host A
Host B
one segment
RTT
two segments
four segments
time
Transport Layer

91. Congestion Window /mss
Refinement
Q: When should the exponential increase switch to linear?
A: When CongWin gets to 1/2 of its value before timeout.
“Fast Recovery”
(after 3 dup. ACKs)
1/2
Congestion Window /mss
“Slow Start”
Implementation:
Variable Threshold
At loss event, Threshold is set to 1/2 of CongWin just before loss event
Transmission Round
(time/rtt)
Transport Layer

92. CongWin <= Threshold: Doubles each RTT (add MSS for each MSS ACKed)
CongWin > Threshold: Adds MSS each RTT
Time Out: Threshold = 1/2 CongWin, CongWin = 1 (Slow-Start)
3-Dup Ack: Threshold = 1/2 CongWin, CongWin = Threshold (Fast Recovery)
CongWin / mss
Time Out
CongWin = 20
3 Dup. ACKs 12 6
Transport Layer
3-96

93. Refinement: inferring loss
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
Philosophy:
3 dup ACKs indicates network capable of delivering some segments
timeout indicates a “more alarming” congestion scenario
Transport Layer

94. Summary: 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 (no slow start, linear mode immediately).
When timeout occurs, Threshold set to CongWin/2 and CongWin is set to 1 MSS.
Transport Layer

95. TCP sender congestion control
State
Event
TCP Sender Action
Commentary
Slow Start (SS)
ACK receipt for previously unacked data
CongWin = CongWin + MSS,
If (CongWin > Threshold)
set state to “Congestion Avoidance”
Resulting in a doubling of CongWin every RTT
Congestion
Avoidance (CA)
CongWin = CongWin+MSS * (MSS/CongWin)
Additive increase, resulting in increase of CongWin by 1 MSS every RTT
SS or CA
Loss event detected by triple duplicate ACK
Threshold = CongWin/2,
CongWin = Threshold,
Set state to “Congestion Avoidance”
Fast recovery, implementing multiplicative decrease. CongWin will not drop below 1 MSS.
Timeout
CongWin = 1 MSS,
Set state to “Slow Start”
Enter slow start
Duplicate ACK
Increment duplicate ACK count for segment being ACKed
CongWin and Threshold not changed
Transport Layer

96. TCP throughput
What’s the average throughout of TCP as a function of window size and RTT?
Ignore slow start
Let W be the window size when loss occurs.
When window is W, throughput is W/RTT
Just after loss, window drops to W/2, throughput to W/2RTT.
Average throughout: .75 W/RTT (assuming loss occurs every time CongWin grows to W)
Transport Layer

97. TCP Futures: TCP over “long, fat pipes”
Example: With 1500 byte segments and 100ms RTT, if we want 10 Gbps throughput:
Requires window size, W = 125,000,000 bytes (for 83, byte in-flight segments)
New versions of TCP for high-speed needed!
Present W of 65,000 bytes -> 5 Mbps
The TCP Window-Multiplier option allows the Window to be incearsed by a exponent of 2.
Transport Layer

98. TCP Fairness
Fairness goal: if K TCP sessions share same bottleneck link of bandwidth R, each should have average rate of R/K
TCP connection 1
bottleneck
router
capacity R
TCP
connection 2
Transport Layer

99. Why is TCP fair? Two competing sessions:
Additive increase gives slope of 1, as throughout increases
multiplicative decrease decreases throughput proportionally
Rmax
Finally: equal bandwidth share
loss: decrease windows by factor of 2
congestion avoidance: additive increase
Connection 2 throughput
R1,R2 -losses: decrease windows by factor of 2
congestion avoidance: additive increase
Start: R1 = 5 x R2, R1 +R2 < Rmax
R1/2,R2/2
Start
Connection 1 throughput
Rmax
Transport Layer

100. Fairness (more) Fairness and UDP Multimedia apps often do not use TCP
do not want rate throttled by congestion control
Instead use UDP:
pump audio/video at constant rate, tolerate packet loss
Research area: make UDP more TCP friendly
Solution: reserve 50% of router buffer space for TCP segments (excess UDP segments dropped).
Fairness and parallel TCP connections
nothing prevents app from opening parallel connections between 2 hosts.
Web browsers do this
Example: link of rate R supporting 9 connections;
new app asks for 1 TCP, gets rate R/10
new app asks for 10 TCPs, gets R/2 !
Browsers using parallel connections are not fair, but work better.
Fairness is not an issue if there is no congestion.
Transport Layer

101. Fixed congestion window (1)
First case:
WS > R * RTT (Rate Limited)
ACK for first segment in window returns before window’s worth of data sent (so no idle time)
Data sent at full rate, R
R * RTT is the Bandwidth-Delay Product
Transport Layer

102. Fixed congestion window (2)
Second case:
WS < R*RTT
(Window Limited)
wait for ACK after sending window’s worth of data sent
Data sent at rate WS/RTT < R
Transport Layer

103. TCP Delay Modeling: Slow Start (2)
Delay components:
2 RTT for connection estab and request
O/R to transmit object
time server idles due to slow start
Server idles: for RTT
O is object size,
R is transmission rate
RTT is round-trip-time.
Transport Layer

104. TCP Delay Modeling (3)
As the CongWin grows, the connection can change from being
"Window Limited" (DataRate=W/RTT) to being
"Transmission Rate Limited" (DataRate = R).
Transport Layer

105. TCP Options source port # dest port # sequence number
32 bits
source port #
dest port #
Options appear in extra bytes appended to the end of the TCP header:
number of bytes,
option ID,
parameter.
A receiver can ignore Options it does not understand (extensible).
sequence number
acknowledgement number
head
len
not
used U A P R S F
Receive window
checksum
Urg data pnter 20
Options (variable length)
<60
data
Common Options
Time Stamp (RFC1323)
Window Scale Option (RFC1323) Multiply Window by 2^n.
SACK (RFC2018) Acknowledge non-contiguous blocks received.
NOP - Pads out options to multiple of 4 bytes

106. Chapter 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”
Transport Layer

107. TCP Connection Management
Three way handshake:
Step 1: client host sends TCP SYN segment to server
specifies initial seq #
TCP option sends client’s MSS*
no data
Step 2: server host receives SYN, replies with SYN-ACK segment
server allocates buffers
specifies server initial seq. #
TCP option sends server’s MSS
Step 3: client receives SYN-ACK, replies with ACK segment, normally has no 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
Socket clientSocket = new Socket("hostname","port number");
server: contacted by client
Socket connectionSocket = welcomeSocket.accept();
MSS - maximum segment size, usually 1420 bytes.
Transport Layer

108. 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
close
closed
timed wait
Or, either host sends a packet with the RES (reset) flag bit set.
This is frequent between Web servers and browsers,
Transport Layer

109. TCP Connection Management (cont)
TCP server
lifecycle
TCP client lifecycle
(client initiates close)
Transport Layer