Lecture 18
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Homework 5, Wireshark Project 3,
Programming Project 3 due today.
Questions?
Thursday, October 18
CS 475 Networks - Lecture 18
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Outline
Chapter 5 - End-to-End Protocols
5.1 Simple Demultiplexer (UDP)
5.2 Reliable Byte Stream (TCP)
5.3 Remote Procedure Call (RPC)
5.4 Transport for Real-Time Applications (RTP)
5.5 Summary
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Introduction
We've covered connecting computers. The
transport layer deals with connecting processes
running on computers.
A transport protocol may be expected to provide:
guaranteed delivery, in-order delivery, no
duplicates, support for large messages,
synchronization, flow control, support for multiple
processes.
Lower layers may: drop messages, reorder
messages, deliver duplicates, limit message size,
deliver after a long delay.
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Simple Demultiplexer (UDP)
UDP extends the host-to-host service of the
network to a process-to-process service but adds
no other functionality.
UDP uses 16-bit port numbers to demultiplex
between processes. (A process is identified with
a port number/IP number pair.)
UDP header format
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Simple Demultiplexer (UDP)
Implementation of the
port abstraction may
vary from OS to OS.
Typically, each port is
associated with a
message queue.
When a process
receives a message
one is removed from
the queue.
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Simple Demultiplexer (UDP)
How does a client know which port to send a
message to on the server? The server may use a
well-known port (see /etc/services on a UNIX
machine).
Alternatively, the server could run a port mapper
process on a well-known port. The client
communicates with the port mapper to find the
port number of the desired process.
UDP does employ a checksum to verify a
message. Packets with errors are dropped.
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Reliable Byte Stream (TCP)
In addition to demultiplexing, TCP provides
guaranteed, reliable, in-order delivery with flow
and congestion control. TCP connections are fullduplex.
Flow control prevents the sender from
overwhelming the receiver. Congestion control
prevents the sender from overwhelming the
network (switches, links).
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End-to-End Issues
The sliding window algorithm used by TCP is like
that used on a point-to-point link (Section 2.5.2),
but there are important differences:
1)Setup (exchange of state so the sliding window algorithm
can start) and teardown are needed
2)RTTs are variable, so timeouts must be adaptive,
3)Packets can be reordered (the maximum segment lifetime
or MSL is typically 120 s)
4)Resources are not tied to a single link and can not be
determined in advance (flow control needed),
5)Congestion is possible (congestion control needed)
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Segment Format
TCP is a byte-oriented protocol. Bytes are
normally collected into segments before before
being sent to the destination.
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Segment Format
The TCP header is
shown at right. A
TCP connection is
identified by the 4tuple (SrcPort,
SrcIPAddr, DstPort,
DstIPAddr).
The HdrLen is the size of the header in 32-bit
words.
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Segment Format
The Acknowledgment, SequenceNum and
AdvertisedWindow fields are used by the sliding
window algorithm. Each transmitted byte has a
corresponding SequenceNum.
Acknowledgment and AdvertisedWindow are
associated with received data.
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Segment Format
The Flags field contains 6 bits: SYN, FIN,
RESET, PUSH, URG, and ACK.
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SYN and FIN are used to set up a connection.
RESET indicates that the receiver is confused and wants
to abort the connection.
PUSH indicates that data should be send immediately.
URG signifies that the segment contains urgent data. The
UrgPtr field contains the number of urgent data bytes.
ACK is set when the Acknowledgment field is valid.
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Connection Establishment
A three-way handshake
is used to set up the
connection. Packets
contain the initial
sequence numbers to
be used by the client
and the server (x and y) in subsequent packets.
The TCP specification requires that the initial
sequence numbers be random numbers.
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Connection Establishment
A trans. diagram for
TCP setup and tear
down is shown at
right. Rectangles
show states. Arcs
have tags of the
form event/action
Retransmissions
due to timeouts are
not shown.
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Sliding Window Revisited
The sliding window algorithm discussed
previously provided reliable, in-order delivery.
TCP's sliding window algorithm extends the prior
one by adding flow control.
Flow control is achieved by having the receiver
advertise a window size to the sender instead of
using a fixed-size window. The sender is limited to
sending no more than AdvertisedWindow bytes
of unacknowledged data at any time.
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Sliding Window Revisited
The sender maintains three pointers where:
LastByteAcked ≤ LastByteSent ≤ LastByteWritten
while on the receiver:
LastByteRead < NextByteExpected ≤ LastByteRcvd + 1
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Sliding Window Revisited
Assume the send and receive buffers are of size
MaxSendBuffer and MaxRcvBuffer. On the
receive side TCP must keep:
LastByteRcvd – LastByteRead ≤ MaxRcvBuffer
The advertised window size is
AdvertisedWindow = MaxRcvBuffer –
((NextByteExpected – 1) – LastByteRead)
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Sliding Window Revisited
On the sending side TCP ensures:
LastByteSent – LastByteAcked ≤ AdvertisedWindow
while maintaining
LastByteWritten – LastByteAcked ≤ MaxSendBuffer
If the sending process tries to write n bytes in
such a way that this inequality would not be
maintained then the process is blocked.
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Sliding Window Revisited
A 32-bit sequence number will wrap around in 57
minutes at a 10 Mbps transmit rate, but in only 36
seconds at 1 Gbps. An extension to TCP extends
the sequence number space.
A 16-bit AdvertisedWindow field allows for a 64
KB window. It should be large enough to allow for
a full delay x BW product. A cross country delay
of 100 ms at 10 Mbps corresponds to 122 KB.
The TCP extension increases the advertised
window size also.
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Triggering Transmission
TCP will transmit a segment when (1) it has
collected a maximum segment size (MSS)
number of bytes, (2) the sending process tells it to
(a push), or (3) a “timer” expires.
Nagle's Algorithm:
when there is data to send
if both the data and the window ≥ MSS
send a full segment
else if there is unACKed data in flight
buffer data until ACK arrives
else
send all data now
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Adaptive Retransmission
Originally, a TimeOut value for retransmission
was computed using:
EstimatedRTT = α EstimatedRTT +
(1 – α)SampleRTT
TimeOut = 2 x EstimatedRTT
where SampleRTT is the time between when a
segment is sent and its ACK arrives.
The original TCP spec recommended a value of α
between 0.8 and 0.9.
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Adaptive Retransmission
Unfortunately an ACK for a retransmission is
identical to an ACK for the original. This can lead
to incorrect values for SampleRTT.
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Adaptive Retransmission
The Karn/Partridge algorithm fixed the problem
quite simply. SampleRTT was measured only for
segments that have been sent once.
The new algorithm included a second change.
After each retransmit the next timeout value would
be set to twice the previous timeout value
(exponential backoff). This helped to alleviate
problems due to network congestion.
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Adaptive Retransmission
The original algorithm did not handle situations in
which the SampleRTT might vary a lot. The
Jackobson/Karels algorithm was an improvement:
Difference = SampleRTT – EstimatedRTT
EstimatedRTT = EstimatedRTT+(δ x Difference)
Deviation = Deviation+δ(|Difference|-Deviation)
TimeOut = μ x EstimatedRTT + φ x Deviation
where μ was typically 1 and φ was 4.
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Record Boundaries
TCP has two features that allow record
boundaries to be put into the byte stream.
TCP allows data to be flagged as urgent or out-ofband. Urgent data can be used to indicate the
end of a record.
A TCP push operation can be used to indicate a
complete record. (The sockets API does not
provide access to the PUSH flag.)
It is usually simpler for record boundary markers
to be inserted by the application.
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TCP Extensions
There have been four optional extensions to TCP
that are implemented using Options in the TCP
header:
1) The sender places a 32-bit time stamp in the
header. The receiver echoes the time stamp in
the ACK. This allows for accurate
measurement of the RTT.
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TCP Extensions
2) The sequence number and the time stamp are
examined to determine if the sequence number
has wrapped around.
3) A scaling factor can be included to advertise a
window larger than 64 KB.
4) The receiver can respond with a selective
acknowledgment (SACK). This allows the
sender to (re)transmit just missing segments.
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Performance
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Now that we have a complete protocol graph,
we can discuss how to measure its
performance as seen by applications.
In particular, as network speeds increase, can
a protocol like TCP provide enough data to
keep the network full?
Simple host-to-host
in a room. 2 2.4GHz
dual cores; 2 Gbps
bandwidth.
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Performance
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TTCP benchmark using
various sizes of
messages.
Note: "perfect" network,
measures TCP
implementation and
workstation
hardware/software only.
Will see other issues
like congestion.
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Alternative Design Choices
TCP is a stream-oriented protocol as opposed to
a request/reply protocol. We will examine a
request/reply protocol (RPC) next time. (TCP can
be used for request/reply applications, but there
are complications.)
TCP is a byte-stream rather than a messagestream service. (Record boundaries can however
be inserted into the byte stream.)
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Alternative Design Choices
TCP uses connection setup and teardown. It is
possible to send all connection parameters with
the first data message. TCP setup allows a
receiver to reject a connection before any data is
sent. TCP teardown means that “keep alive”
messages don't need to be sent.
TCP uses window-based versus rate-based flow
control. There are similarities but also some
interesting differences.
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In-class Exercises
Log on locally under Linux or log on remotely to
csserver to answer the following questions:
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How do we send out-of-band data via TCP? (man send)
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How do we receive out-of-band data? (man recv)
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Which of the four TCP extensions described in class are
supported under Linux? (man tcp)
What acronym is used for the TCP extension that helps to
determine if the sequence number has wrapped around? What
does this acronym stand for?
Is there a way to disable Nagle's algorithm so that segments
are sent immediately? If so, how?
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In-class Exercises
The Linux /proc pseudo-filesystem interface can
be used to tune many of the TCP algorithms.
Changing the parameters requires system
administration privileges. Use cat to examine
appropriate /proc file contents (man tcp) and
determine the answers to the following:
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Is the optional SACK extension enabled?
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What is the default receive buffer size?
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Is the optional window scaling extension enabled?
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What is the default congestion control algorithm? Which
algorithms are available for use?
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