Summary of Last Lecture

1
Summary of Last Lecture

• Transport layer services and protocols
• Logical communication
• End-to-end service
• TCP reliable, in-order delivery
• UDP unreliable, unordered delivery

2
Summary of Last Lecture

• Multiplexing and demultiplexing
• Receiving host demultiplexing
• Sending host multiplexing
• Connectionless demultiplexing
• Identified by two-tuple
• (dest IP, dest port)
• Connection-oriented demultiplexing
• Identified by four-tuple
• (source IP, source port, dest IP, dest port)

3
Summary of Last Lecture

• Roadmap of reliable data transfer
• Rdt 1.0 no bit errors, no loss of packets
• Rdt 2.0 channel with bit errors
• Error detection
• Receiver feedback using control msgs (ACK, NAK)
• Rdt 2.1 handles garbled ACK/NAKs
• Introduction of sequence number
• Rdt 2.2 a NAK-free protocol
• Duplicate ACK at sender retransmit current pkt

4
Summary of Last Lecture

• Roadmap of reliable data transfer (cont.)
• Rdt 3.0 channels with errors and pkt loss
• Sequence number
• Timer

5
Summary of Last Lecture

• Go-Back-N
• Sender
• K-bit sequence number
• Cumulative ACK
• Timer
• Timeout(n) retransmit pkt n and all higher
  sequence number pkts in window
• Receiver
• No receiver buffering
• Repeat ACK pkt with highest in-order sequence
  number

6
Summary of Last Lecture

• Selective Repeat
• Sender
• Only retransmit pkts for which ACK not received
• Receiver
• Buffers out-of-order pkts
• Individually acknowledges all correctly received
  pkts

7
CS352- TCP and UDP

• Dept. of Computer Science
• Rutgers University

8
The Internet Transport Layer

• Two transport layer protocols supported by the
  Internet
• Reliable
• The Transport Control Protocol (TCP)
• Unreliable
• The Unreliable Datagram Protocol (UDP)

9
UDP

• UDP is an unreliable transport protocol that can
  be used in the Internet
• UDP does not provide
• connection management
• flow or error control
• guaranteed in-order packet delivery
• UDP is almost a null transport layer

10
Why UDP?

• No connection needs to be set up
• Throughput may be higher because UDP packets are
  easier to process, especially at the source
• The user doesnt care if the data is transmitted
  reliably
• The user wants to implement his or her own
  transport protocol

11
UDP 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
12
UDP checksum

• Goal detect errors (e.g., flipped bits) in
  transmitted segment

• 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 .
• No action to recover from an error

• Sender
• treat segment contents as sequence of 16-bit
  integers
• checksum 1s complement of (1s complement sum
  of segment contents)
• sender puts checksum value into UDP checksum
  field

13
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 1 1 1 0 0 1 1 0 0 1 1 0 0 1 1
0 1 1 1 0 1 0 1 0 1 0 1 0 1 0 1 0
1 1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1
1 1 1 0 1 1 1 0 1 1 1 0 1 1 1 1
0 0 1 0 1 0 0 0 1 0 0 0 1 0 0 0 0
1 1
Wraparound the carry
sum
Checksum (complement)
14
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) inits
  sender, receiver state before data exchange
• flow controlled
• sender will not overwhelm receiver

15
TCP

• TCP provides the end-to-end reliable connection
  that IP alone cannot support
• The protocol
• Frame format
• Connection management
• Retransmission
• Flow control
• Congestion control

16
TCP 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 estab (setup,
teardown commands)
Internet checksum (as in UDP)
17
TCP Frame Fields

• Source Destination Ports
• 16 bit port identifiers for each packet
• Sequence number
• The packets unique sequence ID
• Acknowledgement number
• The sequence number of the next packet expected
  by the receiver

18
TCP Frame Fields (contd)

• Window size
• Specifies how many bytes may be sent after the
  first acknowledged byte
• Checksum
• Checksums the TCP header and IP address fields
• Urgent Pointer
• Points to urgent data in the TCP data field

19
TCP Frame Fields (contd)

• Header bits
• URG Urgent pointer field in use
• ACK Indicates whether frame contains
  acknowledgement
• PSH Data has been pushed. It should be
  delivered to higher layers right away.
• RST Indicates that the connection should be
  reset
• SYN Used to establish connections
• FIN Used to release a connection

20
TCP 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
Host A
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
21
TCP Connection Establishment

• Three-way Handshake

Host A
Host B
SYN (seqx)
SYN (seqy, ACKx1)
SYN (seqx1, ACKy1)
22
TCP Connection Tear-down

• Two double handshakes

Host A
Host B
FIN (seqx)
ACK (ACKx1)
A-gtB torn down
FIN (seqy)
ACK (ACKy1)
B-gtA torn down
23
TCP Retransmission

• When a packet remains unacknowledged for a period
  of time, TCP assumes it is lost and retransmits
  it
• TCP tries to calculate the round trip time (RTT)
  for a packet and its acknowledgement
• From the RTT, TCP can guess how long it should
  wait before timing out

24
TCP 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 several recent measurements, not just
  current SampleRTT

• Q how to set TCP timeout value?
• longer than RTT
• but RTT varies
• too short premature timeout
• unnecessary retransmissions
• too long slow reaction to segment loss

25
Round Trip Time (RTT)
Time for data to arrive
Network
Time for ACK to return
RTT Time for packet to arrive at destination
Time for ACK to return from
destination
26
RTT Calculation
Receiver
Sender
0.9 sec
RTT
ACK 2048
2.2 sec
RTT 2.2 sec - 0.9 sec. 1.3 sec
27
Smoothing the RTT measurement

• First, we must smooth the round trip time due to
  variations in delay within the network
• SRTT a SRTT (1-a) RTTarriving ACK
• The smoothed round trip time (SRTT) weights
  previously received RTTs by the a parameter
• a is typically equal to 0.875

28
Retransmission Timeout Interval (RTO)

• The timeout value is then calculated by
  multiplying the smoothed RTT by some factor
  (greater than 1) called b
• Timeout b SRTT
• This coefficient of b is included to allow for
  some variation in the round trip times.

29
Example
Initial SRTT 1.50 a 0.875, b 4.0
RTT Meas.
SRTT
Timeout
1.5 s
1.50
b1.50 6.00
1.0 s
1.50a 1.0(1- a) 1.44
b1.44 5.76
2.2 s
1.44a 2.2(1- a) 1.54
b1.54 6.16
1.0 s
1.54a 1.0(1- a) 1.47
b1.47 5.88
0.8 s
1.47a 0.8(1- a) 1.39
b1.39 5.56
3.1 s
2.0 s
30
Problem with RTT Calculation
Receiver
Sender
Sender Timeout
RTT?
ACK 2048
RTT?
31
Karns Algorithm

• Retransmission ambiguity
• Measure RTT from original data segment
• Measure RTT from most recent segment
• Either way there is a problem in RTT estimate
• One solution
• Never update RTT measurements based on
  acknowledgements from retransmitted packets
• Problem Sudden change in RTT can cause system
  never to update RTT
• Primary path failure leads to a slower secondary
  path

32
Karns algorithm

• Use back-off as part of RTT computation
• Whenever packet loss, RTO is increased by a
  factor
• Use this increased RTO as RTO estimate for the
  next segment (not from SRTT)
• Only after an acknowledgment received for a
  successful transmission is the timer set to new
  RTT obtained from SRTT

33
Another Problem with RTT Calculation

• RTT measurements can sometimes fluctuate severely
• smoothed RTT (SRTT) is not a good reflection of
  round-trip time in these cases
• Solution Use Jacobson/Karels algorithm
• Error RTT - SRTT
• SRTT SRTT (a Error)
• Dev Dev h(Error - Dev)
• Timeout SRTT (b Dev)

34
Jacobson/Karels AlgorithmExample
Error RTT - SRTT SRTT SRTT (a Error) Dev
Dev d (Error - Dev) Timeout SRTT (b
Dev)
Initial SRTT 1.50, Dev 0 a 0.125, d 0.25,
b 4.0
RTT Meas.
SRTT
Error
Dev.
Timeout
1.5 s
1.50
0.0
0.00
1.50
1.0 s
1.44
-0.50
0.13
1.94
2.2 s
1.54
0.76
0.28
2.67
1.0 s
1.47
-0.54
0.35
2.85
0.8 s
1.39
-0.67
0.43
3.09
3.1 s
2.0 s
35
Example RTT computation
36
TCP Flow Control

• TCP uses a modified version of the sliding window
• In acknowledgements, TCP uses the Window size
  field to tell the sender how many bytes it may
  transmit
• TCP uses bytes, not packets, as sequence numbers

37
TCP Flow Control (contd)
Important information in TCP/IP packet headers
Number of bytes in packet (N)
Sequence number of first data byte in packet (SEQ)
N
SEQ
Send
Window size at the receiver (WIN)
Sequence number of next expected byte (ACK)
ACK bit set
ACK
WIN
Recv
Contained in TCP header
Contained in IP header
38
Example TCP session

• (1)remus tcpdump -S host scullyKernel
  filter, protocol ALL, datagram packet
  sockettcpdump listening on all devices
• 151522.152339 eth0 gt remus.4706 gt
  scully.echo S 12642965041264296504(0) win 32120
  ltmss 1460,sack OK,timestamp 71253512 0,nop,wscale
  0gt 151522.153865 eth0 lt scully.echo gt
  remus.4706 S 875676030875676030(0) ack
  1264296505 win 8760 ltmss 1460gt151522.153912
  eth0 gt remus.4706 gt scully.echo .
  12642965051264296505(0) ack 875676031 win 32120

remus telnet scully 7 A ltreturngt A
39
Example TCP session
Timestamp
Source IP/port
Dest IP/port
Packet 1 151522.152339 eth0 gt remus.4706 gt
scully.echo S 12642965041264296504(0) win 32120
ltmss 1460,sackOK,timestamp 71253512 0,nop,wscale
0gt (DF)
Flags
Packet 2 151522.153865 eth0 lt scully.echo gt
remus.4706 S 875676030875676030(0) ack
1264296505 win 8760 ltmss 1460)
Options
Packet 3 151522.153912 eth0 gt remus.4706 gt
scully.echo . 12642965051264296505(0) ack
875676031 win 32120
Window
Start Sequence Number
Acknowledgement Number
End Sequence Number
40
TCP data transfer
Packet 4151528.591716 eth0 gt remus.4706 gt
scully.echo P 12642965051264296508(3) ack
875676031 win 32120
data
Packet 5 151528.593255 eth0 lt scully.echo gt
remus.4706 P 875676031875676034(3) ack
1264296508 win 8760
bytes
41
TCP Flow Control (contd)
Receivers buffer
Receiver
Sender
Application does a 2K write
0
4K
Empty
ACK 2048 WIN 2048
Application does a 3K write
Full
Sender is blocked
Application reads 2K
ACK 4096 WIN 0
ACK 4096 WIN 2048
Sender may send up to 2K
42
TCP Flow Control (contd)
Piggybacking Allows more efficient
bidirectional communication
Data from A to B
ACK for data from B to A
N
SEQ
ACK
WIN
A
B
N
SEQ
ACK
WIN
Data from B to A
ACK for data from A to B
43
TCP Congestion Control

• Recall Network layer is responsible for
  congestion control
• However, TCP/IP blurs the distinction
• In TCP/IP
• the network layer (IP) simply handles routing and
  packet forwarding
• congestion control is done end-to-end by TCP

44
Self-Clocking Model
Bottleneck link
Pr
Fast link
Pb
1. Send Burst
2. Receive data packet
5. Send a data packet
Data
Receiver
Sender
Acks
4. Receive Acknowledgement
3. Send Acknowledgement
Ab
Ar
Ar
Given Pb Pr Ar Ab Ar (in units of time)
Sending a packet on each ACK keeps the
bottleneck link busy
45
Changing bottleneck bandwidth

• one router, finite buffers
• sender retransmission of lost packet

Host A
lout
lin original data
l'in original data, plus retransmitted data
Host B
finite shared output link buffers
46
TCP Congestion Control

• Goal achieve self-clocking state
• Even if dont know bandwidth of bottleneck
• Bottleneck may change over time
• Two phases to keep bottleneck busy
• Slow-start ramps up to the bottleneck limit
• Packet loss signals we passed bandwidth of
  bottleneck
• Congestion Avoidance tries to maintain self
  clocking mode once established

47
TCP Congestion Window

• TCP introduces a second window, called the
  congestion window
• This window maintains TCPs best estimate of
  amount of outstanding data to allow in the
  network to achieve self-clocking

48
TCP Congestion Window

• To determine how many bytes it may send, the
  sender takes the minimum of the receiver window
  (i.e. sliding window) and the congestion window
• Example
• If the receiver window says the sender can
  transmit 8K, but the congestion window is only
  4K, then the sender may only transmit 4K
• If the congestion window is 8K but the receiver
  window says the sender can transmit 4K, then the
  sender may only transmit 4K

49
TCP Slow Start Phase

• TCP defines the maximum segment size as the
  maximum size a TCP packet can be (including
  header)
• TCP Slow Start
• Congestion window starts small, at 1 segment size
• Each time a transmitted segment is acknowledged,
  the congestion window is increased by one maximum
  segment size
• On each ack, cwndcwnd 1

50
TCP Slow Start (contd)
Congestion Window Size
Event

1K A sends 1 segment to B B ACKs the
segment 2K A sends 2 segments to B B ACKs both
segments 4K A sends 4 segments to B B ACKs all
four segments 8K A sends 8 segments to B B
ACKs all eight segments 16K and so on
51
TCP Slow Start (contd)

• Congestion window size grows exponentially (i.e.
  it keeps on doubling)
• Packet losses indicate congestion
• Packet losses are determined by using timers at
  the sender
• When a timeout occurs, the congestion window is
  reduced to one maximum segment size and
  everything starts over

52
TCP Slow Start

• When connection begins, increase rate
  exponentially until first loss event
• double CongWin every RTT
• done by incrementing CongWin for every ACK
  received
• Summary initial rate is slow but ramps up
  exponentially fast

Host A
Host B
one segment
RTT
two segments
four segments
53
TCP Slow Start (contd)
Timed out Transmissions
Congestion window
Transmission Number
1 Maximum Segment Size
54
TCP Slow Start (contd)

• TCP Slow Start by itself is inefficient
• Although the congestion window builds
  exponentially, it drops to 1 segment size every
  time a packet times out
• This leads to low throughput

55
TCP Linear Increase Threshold

• Establish a threshold at which the rate increase
  is linear instead of exponential to improve
  efficiency
• Algorithm
• Start the threshold at 64K (ssthresh)
• Slow start
• Once the threshold is passed, only increase the
  congestion window size by 1 segment size for each
  congestion window of data transmitted
• For each ack received, cwnd cwnd
  (mssmss)/cwnd
• If a timeout occurs, reset the congestion window
  size to 1 segment and set threshold to
  max(2mss,1/2 of MIN(sliding window, congestion
  window))

56
TCP Linear Increase Threshold Phase
Example Maximum segment size 1K Assume
SSthresh32K
Timeout occurs when MIN(sliding window,
congestion window) 40K
Congestion window
Thresholds
40K
32K
20K
1K
Transmission Number
57
TCP Fast Retransmit

• Another enhancement to TCP congestion control
• Idea When sender sees 3 duplicate ACKs, it
  assumes something went wrong
• The packet is immediately retransmitted instead
  of waiting for it to timeout
• Why?
• Note that acks sent by the receiver when it
  receives a packet
• Dup ack implies something is getting through
• Better than time out

58
TCP Fast RetransmitExample
Receiver
Sender
MSS 1K
ACK of new data
ACK 2048 WIN 31K
Duplicate ACK 1
ACK 2048 WIN 30K
Duplicate ACK 2
ACK 2048 WIN 29K
Fast Retransmit occurs (2nd packet is
now retransmitted w/o waiting for it to timeout)
Duplicate ACK 3
ACK 2048 WIN 28K
ACK 2048 WIN 27K
ACK 7168 WIN 26K
59
TCP Fast Recovery

• Yet another enhancement to TCP congestion control
• Idea Dont do a slow start after a fast
  retransmit
• Instead, use this algorithm
• Drop threshold to max(2mss,1/2 of MIN(sliding
  window, congestion window))
• Set congestion window to threshold 3 MSS
• For each duplicate ACK (after the fast
  retransmit), increment congestion window by MSS
• When next non-duplicate ACK arrives, set
  congestion window equal to the threshold

60
TCP Fast RecoveryExample
Sender
SW29K,TH15K, CW20K
Continuing with the Fast Retransmit Example...
SW28K,TH15K, CW20K
ACK 2048 WIN 28K
Fast Retransmit Occurs
MSS1K Sliding Window (SW) Congestion Threshold
(TH) Congestion Window (CW)
SW28K, TH10K, CW13K
ACK 2048 WIN 27K
SW27K, TH10K, CW14K
ACK 7168 WIN 26K
SW26K, TH10K, CW10K
61
Resulting TCP Sawtooth
In steady state, window oscillates around the
bottlenecks capacity (I.e. number of
outstanding bytes in transit)
Congestion window
Slow Start
Bottleneck Capacity
Linear Mode
40K
Sawtooth
32K
20K
1K
Transmission Number
62
TCP Recap

• Timeout Computation
• Timeout is a function of 2 values
• the weighted average of sampled RTTs
• The sampled variance of each RTT
• Congestion control
• Goal Keep the self-clocking pipe full in spite
  of changing network conditions
• 3 key Variables
• Sliding window (Receiver flow control)
• Congestion window (Sender flow control)
• Threshold (Senders slow start vs. linear mode
  line)

63
TCP Recap (cont)

• Slow start
• Add 1 segment for each ACK to the congestion
  window
• -Doubles the congestion windows volume each
  RTT
• Linear mode (Congestion Avoidance)
• Add 1 segments worth of data to each congestion
  window
• Adds 1 segment per RTT

64
Algorithm 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 max(FlightSize/2,2mss) and CongWin set to
  Threshold3mss. (Fast retransmit, Fast recovery)
• When timeout occurs, Threshold set to
  max(FlightSize/2,2mss) and CongWin is set to 1
  MSS.
• FlightSize MIN(sliding window, congestion
  window) - The amount of data that has been sent
  but not yet acknowledged.

65
TCP sender congestion control
66
TCP Fairness

• Fairness goal if K TCP sessions share same
  bottleneck link of bandwidth R, each should have
  average rate of R/K

67
Why is TCP fair?

• Two competing sessions
• Additive increase gives slope of 1, as throughout
  increases
• multiplicative decrease decreases throughput
  proportionally

R
equal bandwidth share
loss decrease window by factor of 2
Connection 2 throughput
congestion avoidance additive increase
Connection 1 throughput
R
68
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 TCP friendly

• 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 11 TCPs, gets R/2 !

69
Delay modeling

• Q How long does it take to receive an object
  from a Web server after sending a request?
• Ignoring congestion, delay is influenced by
• TCP connection establishment
• data transmission delay
• slow start

• Notation, assumptions
• Assume one link between client and server of rate
  R
• S MSS (bits)
• O object size (bits)
• no retransmissions (no loss, no corruption)
• Window size
• First assume fixed congestion window, W segments
• Then dynamic window, modeling slow start

70
Fixed congestion window (1)

• First case
• WS/R gt RTT S/R ACK for first segment in window
  returns before windows worth of data sent

delay 2RTT O/R
71
Fixed congestion window (2)

• Second case
• WS/R lt RTT S/R wait for ACK after sending
  windows worth of data sent

delay 2RTT O/R (K-1)RTT (W-1)S/R
delay 2RTT O/R (K-1)S/R RTT - WS/R
K O/WS
72
TCP Delay Modeling Slow Start (1)

• Now suppose window grows according to slow start
  
• Will show that the delay for one object is

where P is the number of times TCP idles at
server
- where Q is the number of times the server
idles if the object were of infinite size. -
and K is the number of windows that cover the
object.
73
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 P minK-1,Q times

• Example
• O/S 15 segments
• K 4 windows
• Q 2
• P minK-1,Q 2
• Server idles P2 times

74
TCP Delay Modeling (3)
75
TCP Delay Modeling (4)
Recall K number of windows that cover
object How do we calculate K ?
Calculation of Q, number of idles for
infinite-size object, is similar (see HW).