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COT 4600 Operating Systems Fall 2009
Dan C. Marinescu
Office: HEC 439 B
Office hours: Tu-Th 3:00-4:00 PM
Lecture 27
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Schedule
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Tuesday November 24 - Project phase 4 and HW 6 are due
 Tuesday December 1st -Research projects instead of final exam presentation
 Thursday December 3rd - Class review
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Last time:
 More on Scheduling
 Network properties - (Chapter 7) - available online from the publisher of
the textbook
 Layers
 Data Link Layer
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Today:
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Transport protocols; end-to-end problems
Next Time:
 Project discussion
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2
Requirements of application level-processes.
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The transport protocols are expected to:
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Guarantee message delivery
Deliver messages in the same order they are sent
Deliver at most one copy of each message
Support arbitrarily large messages
Support synchronization between sender and receiver
Allow the receiver to apply flow control to the sender
Support multiple application processes on each host
Best-effort networks
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Services provided by best-effort networks
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Drop messages
Re-order messages
Deliver duplicate copies of a given message
Limit messages to some finite size
Deliver messages after an arbitrarily long delay
Challenges  develop algorithms that turn the less-thandesirable properties of the underlying network into the high level
services required by application programs.
UDP
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Provides a process-to-process transport service.
Unreliable and unordered datagram service
No flow control
Multiplexing
Pseudo-header: fields from the IP header (1)+(2)+(3)
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(1) protocol number
 (2) source IP address
 (3) destination IP address
 UDP length field
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UDP Checksum optional in IPv4 but mandatory in IPv6
 pseudo header + udp header + data
UDP ports
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Endpoints identified by ports.
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servers run at well-known ports (e.g. Web server at port 80, Domain
Server at port 53)
see /etc/services on Unix
UDP header
Reliable service - TCP
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Connection-oriented
Byte-stream
 sending process writes some number of bytes
 TCP breaks into segments and sends via IP
 receiving process reads some number of bytes
Full duplex
Flow control: keep sender from overrunning receiver
Congestion control: keep sender from overrunning network
End-to-end issues
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TCP is based on sliding window protocol used at data link
level, but the situation is very different.
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Potentially connects many different hosts need explicit
connection establishment and termination
Potentially different RTT need adaptive timeout mechanism
Potentially long delay in network  need to be prepared for arrival
of very old packets
Potentially different capacity at destination need to
accommodate different amounts of buffering
Potentially different network capacity  need to be prepared for
network congestion
End-to-end issues (cont’d)
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At a link level congestion is visible in the queue of packets at the
sender. Network congestion much tougher to cope with, detect and
then prevent.
TCP uses the sliding window algorithm on an end-to end basis to
provide reliable/ordered delivery.
X.25 uses the sliding window protocol on a hop-by-hop basis.
A sequence of hop-by-hop guarantees does not add up to an endto-end guarantee.
End-to-end argument: a function should not be provided at a lower
levels of a system unless it can be completely and correctly
implemented at that level.
Yet for performance optimization a function may be incompletely
provided at a low level. E.g. error detection is provided on a hop by
hop basis. Why send all the way a corrupted packet?
Segment Format
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Each connection identified with 4-tuple:
 <SrcPort, SrcIPAddr, DstPort, DstIPAddr>
Sliding window + flow control
 Acknowledgment, SequenceNum,
AdvertisedWindow
Flags: SYN, FIN, RESET, PUSH, URG, ACK
Checksum: pseudo header + tcp header + data
The state transition diagram of a TCP connection
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Two types of events trigger a transition:
 a segment arrives from the peer
 the local application process invokes and operation on TCP
Lecture 27
12
Connection establishment and termination
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Connection establishment is an asymmetric activity:
 A server does a passive open.
 A client does an active open
 The two parties have to agree upon on a set of parameters the
starting numbers the two sides plan to use for their respective byte
streams.
 Three-Way Handshake
Connection termination is symmetric, each side has to close the
connection independently. One side can do a close, meaning that it can
no longer send the data, but the other side may keep the other half of the
connection open and continue sending data.
Open a connection
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The information exchanged during the three-way
handshake
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SN - initial sequence number the client wants to use
 SN - initial sequence number the server wants to use
 SYN  flag indicating a sequence number is sent
 SYN+ACK  flags indicating a sequence number is sent and the
next sequence number expected (ACK).
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The three-ways handshake:
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a client does an active open, sends a SYN and moves to
SYN_SENT state.
 when SYN arrives at server, the server moves to SYN_RECVD
state and sends a segment with SYN+ACK flags on.
 when the client receives the segment with SYN+ACK flags on it
moves to ESTABLISHED state
Lecture 27
15
Close a connection
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Three combinations of transitions to close a connection:
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This side closes first:
ESTABLISHED FIN_WAIT_1FIN_WAIT_2
TIME_WAITCLOSED
 The other side closes first:
ESTABLISHEDCLOSE_WAITLAST_ACKCLOSED
 Both sides close at the same time:
ESTABLISHEDFIN_WAIT_1CLOSINGTIME_WAITCLOSING
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It takes twice the maximum life time of an IP datagram in the Internet
( 120 sec) to move from TIME_WAIT to CLOSED.
Reason: the local side of the connection sent an ACK segment in
response to a FIN segment but does not know if the ACK segment
reached the other side. The other side may have retransmitted the
FIN that might arrive late and be acted upon by a new reincarnation
of the connection.
Sliding-window revisited
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Each byte has a sequence number. ACKs are cumulative
Sending side
 LastByteAcked  LastByteSent
 LastByteSent  LastByteWritten
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bytes between LastByteAcked and LastByteWritten must be
buffered
Receiving side
 LastByteRead < NextByteExpected
 bytes between LastByteRead and LastByteRcvd must be buffered
Flow control
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Sender buffer size: MaxSendBuffer
Receive buffer size: MaxRcvBuffer
Receiving side:
 LastByteRcvd - NextByteRead  MaxRcvBuffer
 AdvertisedWindow = MaxRcvBuffer - (LastByteRcvd NextByteRead)
Flow control (cont’d)
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Sending side
NextByteExpected  LastByteRcvd + 1
LastByteSent - LastByteAcked  AdvertisedWindow
EffectiveWindow = AdvertisedWindow –
(LastByteSent - LastByteAcked)
LastByteWritten - LastByteAcked  MaxSendBuffer
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Block sender if
(LastByteWritten - LastByteAcked) + y > MaxSendBuffer
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Always send ACK in response to an arriving data segment
Persist when AdvertisedWindow=0
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Protecting against wraparound
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A byte with a sequence number x may be sent at one time and then on the
same connection a byte with the same sequence number x may be sent
again.
Wrap Around: controlled by the 32-bit SequenceNum
The maximum lifteime of an IP datagram is 120 sec thus we need to have a
wraparound time at least 120 sec.
For slow links OK but no longer sufficient for optical networks. Bandwidth &
Time Until Wrap Around
Bandwidth
T1 (1.5Mbps)
Ethernet (10Mbps)
T3 (45Mbps)
FDDI (100Mbps)
STS-3 (155Mbps)
STS-12 (622Mbps)
STS-24 (1.2Gbps)
Time Until Wrap Around
6.4 hours
57 minutes
13 minutes
6 minutes
4 minutes
55 seconds
28 seconds
Keeping the pipe full
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The SequenceNum, the sequence number space (32 bits) should
be twice as large as the window size (16 bits). It is.
The window size (the number of bytes in transit) is given by the
AdvertisedWindow field (16 bits).
The higher the bandwidth the larger the window size to keep the
pipe full.
Essentially we regard the network as a storage system and the
amount of data is equal to: ( bandwidth x delay )
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Required window size for a 100 msec RTT.
Bandwidth
T1 (1.5Mbps)
Ethernet (10Mbps)
T3 (45Mbps)
FDDI (100Mbps)
STS-3 (155Mbps)
STS-12 (622Mbps)
STS-24 (1.2Gbps)
Delay x Bandwidth Product
18KB
122KB
549KB
1.2MB
1.8MB
7.4MB
14.8MB
Adaptive retransmission –original algorithm
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Measure SampleRTT for each segment/ACK pair
Compute weighted average of RTT
 EstimatedRTT = a x EstimatedRTT + b x SampleRTT
 where a + b = 1
 a between 0.8 and 0.9
 b between 0.1 and 0.2
Set timeout based on EstimatedRTT
 TimeOut = 2 x EstimatedRTT
Karn/Partridge algorithm
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Do not sample RTT when retransmitting
Double timeout after each retransmission
Jacobson/Karels algorithm
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New calculation for average RTT
Diff = SampleRTT - EstimatedRTT
EstimatedRTT = EstimatedRTT + (d x )
Deviation = Deviation + d(|Diff|- Deviation)
 where d is a fraction between 0 and 1
Consider variance when setting timeout value
 TimeOut = m x EstimatedRTT + f x Deviation
 where m = 1 and f = 4
Notes
 algorithm only as good as granularity of clock (500ms on Unix)
 accurate timeout mechanism important to congestion control (later)
Congestion control and resource allocation
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Network resources: link bandwidth and router buffer space.
Global optimization problem - hence very hard.
Flow control versus congestion control (should be fair).
Network model:
 a) Packet switched network. Congestion is often unavoidable.
Congestion control and resource allocation
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b) Connectionless Flows.
Datagram (no state) | Virtual Circuit (hard state) |Flows (soft state)
Flow  a sequence of packets between a source/destination pair following the
same route through the network. Established explicitly or implicitly.
Soft state  cannot explicitly be created and removed by signaling
Congestion control and resource allocation
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c) Underlying service model
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best-effort (assume for now)
multiple qualities of service (later)
Taxonomy of resource allocation mechanisms:
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(1) Router Centric/Host Centric
(2) Reservation Based / Feedback Based
(3) Window based / Rate Based
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Evaluation
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fairness
power (ratio of throughput to delay)
Fairness criteria
Queuing disciplines
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First-In-First-Out (FIFO) with tail drop
 does not discriminate between traffic sources
Priority Queuing
Fair Queuing (FQ)
 explicitly segregates traffic based on flows, one queue per flow
 ensures no flow captures more than its share of capacity
Weighted fair queuing (WFQ). How many bits to transmit each time the
router servers a queue.
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Problem: packets not all the same length
 really want bit-by-bit round robin
 not feasible to interleave bits (schedule on packet basis)
 simulate by determining when packet would finish
For a single flow
 suppose clock ticks each time a bit is transmitted
 let Pi denote the length of packet i
 let Si denote the time when start to transmit packet i
 let Fi denote the time when finish transmitting packet i
 Fi = Si + Pi
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When does router start transmitting packet i
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If before router finished packet i-1 from this flow, then immediately
after last bit of i--1 (Fi-1)
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If no current packets for this flow, then start transmitting when
arrives (call this Ai)
Thus: Fi = MAX(Fi-1, Ai) + Pi
For multiple flows
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calculate Fi for each packet that arrives on each flow
treat all Fi's as timestamps
next packet to transmit is one with lowest timestamp
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Not perfect: can't preempt the packet currently being transmitted
Example
TCP congestion control
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In late 1980’s Van Jacobson proposed to augment TCP with a congestion
control mechanism.
Each source attempts to estimate how much capacity is available in the
network. The source uses:
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(a) the implicit feedback provided by the acknowledgments to determine when
it is safe to insert a new packet into the network (self-clocking), and
 (b) timeouts to detect congestion.
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Three inter-related mechanisms:
Additive Increase/Multiplicative Decrease  new state variable a window that
combines flow control and congestion control.
 Slow Start  mechanism to allow a connection to achieve its window fast.
 Fast Retransmit  a heuristic to deal with timeouts.
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Additive Increase/Multiplicative Decrease
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Objective: adjust to changes in the available capacity
New state variable per connection that limits how much data source has in
transit : CongestionWindow
MaxWin = MIN(CongestionWindow, AdvertisedWindow)
EffWin = MaxWin - (LastByteSent - LastByteAcked)
Idea:
increase CongestionWindow when congestion goes down
 decrease CongestionWindow when congestion goes up
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Question: how does the source determine whether or not the network is
congested?
Answer: a timeout occurs
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timeout signals that a packet was lost
 packets are seldom lost due to transmission error
 lost packet implies congestion
Additive Increase/Multiplicative Decrease
Algorithm:
 increment CongestionWindow by one packet per RTT (linear increase)
 divide CongestionWindow by two whenever a timeout occurs
(multiplicative decrease)
In practice: increment a little for each ACK. MSS – maximum segment size.
 Increment = MSS * MSS/CongestionWindow)
 CongestionWindow += Increment
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Packets in transit during additive increase with one packet
being added each RTT
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Example trace: saw-tooth behavior
Slow start
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Objective: determine the available capacity in the first place
It takes too long to ramp up a connection when starting from scratch.
Idea: double the number of packets TCP
has in transit every RTT.
begin with CongestionWindow = 1 packet
 double CongestionWindow each RTT
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(increment by 1 packet for each ACK)
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Exponential growth, but slower than all
in one blast.
Used...
 when first starting connection
 when connection goes dead waiting for a timeout
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Example trace
Linear growth till about 0.4 sec. Then packets get lost and the window ramps up. A
timeout occurs at 2 sec. Then the window size becomes about half of what it was
before 17 kB versus 34, and increases linearly towards that value…..
Why packets get lost: assume that the network would only support 20 packets from this
source. When they get to the destination and ACKs are sent, the window size is
doubled to 40. Obviously the other half is lost.
Fast retransmit and fast recovery
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Problem: coarse-grain TCP timeouts lead to idle periods
Fast retransmit: use duplicate ACKs to trigger retransmission
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Results
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Fast recovery: remove the slow start phase; go directly to half
the last successful CongestionWindow
Congestion avoidance mechanisms
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TCP's strategy
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to control congestion once it happens
 to repeatedly increase load in an effort to find the point at which
congestion occurs, and then back off
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Alternative strategy
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predict when congestion is about to happen, and reduce the rate at
which hosts send data just before packets start being discarded
 we call this congestion avoidance, to distinguish it from congestion
control
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Two possibilities
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router-centric: DECbit and RED Gateways
 host-centric: TCP Vegas