CS 457 - Lecture 20 Transport Layer: UDP and TCP: Fall 2011

CS 457 – Lecture 20
Transport Layer: UDP and TCP

Fall 2011
Topics
•  Principles underlying transport-layer
services
–  Demultiplexing
–  Detecting corruption
–  Reliable delivery
–  Flow control
•  Transport-layer protocols
–  User Datagram Protocol (UDP)
–  Transmission Control Protocol (TCP)
Role of Transport Layer
•  Application layer
–  Communication between networked applications
–  Protocols: HTTP, FTP, NNTP, and many others
•  Transport layer
–  Communication between processes (e.g.,
socket)
–  Relies on network layer and serves the
application layer
–  Protocols: TCP and UDP
•  Network layer
–  Communication between nodes
–  Protocols: IP
Transport Protocols
•  Provide logical
communication between
application processes running application
transport
on different hosts network
data link network
•  Run on end hosts physical
network
data link
physical
data link
–  Sender: breaks application physical
network
messages into segments, data link
physical network
and passes to network data link
physical
layer network
data link
–  Receiver: reassembles physical
segments into messages, application
passes to application layer transport

network
data link
•  Multiple transport protocol physical
available to applications
–  Internet: TCP and UDP
Internet Transport Protocols
•  Datagram messaging service (UDP)
–  No-frills extension of “best-effort” IP
–  Just send the data – each send is a message
•  Reliable, streaming, in-order delivery (TCP)
–  Connection set-up
–  Discarding of corrupted packets
–  Retransmission of lost packets
–  Flow control
–  Congestion control (next lecture)
•  Services not available
–  Delay guarantees
–  Bandwidth guarantees
Multiplexing and Demultiplexing
•  Host receives IP datagrams
32 bits
–  Each datagram has source
source port # dest port #
and destination IP address,
–  Each datagram carries one
other header fields
transport-layer segment
–  Each segment has source
and destination port application
number data
•  Host uses IP addresses and (message)
port numbers to direct the
segment to appropriate socket
TCP/UDP segment format
User Datagram Protocol (UDP)
•  Lightweight communication between processes

–  Avoid overhead and delays of ordered, reliable delivery
–  Send messages to and receive them from a socket
•  Lightweight delivery service
–  IP plus port numbers to support (de)multiplexing
–  Optional error checking on the packet contents
SRC port DST port
checksum length
DATA
Why Would Anyone Use UDP?
•  Finer control over what data is sent and when
–  As soon as an application process writes into the socket
–  … UDP will package the data and send the packet
•  Low delay
–  UDP just blasts away without any formal preliminaries
–  … which avoids introducing delays such as setup
•  No connection state
–  No allocation of buffers, parameters, sequence #s, etc.
–  … making it easier to handle many active clients
•  Small packet header overhead
–  UDP header is only eight-bytes long
Popular Applications That Use UDP
•  Multimedia streaming
–  Retransmitting lost/corrupted packets is not worthwhile
–  By the time the packet is retransmitted, it’s too late
–  E.g., telephone calls, video conferencing, gaming
•  Simple query protocols like Domain Name System
–  Overhead of connection establishment is overkill
–  Easier to have application retransmit if needed
“Address for www.cnn.com?”
“12.3.4.15”
Transmission Control Protocol (TCP)
•  Connection oriented
– Explicit set-up and tear-down of TCP session
•  Stream-of-bytes service
– Sends and receives a stream of bytes, not messages
– Similar to file I/O
•  Reliable, in-order delivery
– Checksums to detect corrupted data
– Acknowledgments & retransmissions for reliable
delivery
– Sequence numbers to detect losses and reorder data
•  Flow control
–  Prevent overflow of the receiver’s buffer space
•  Congestion control
– Adapt to network congestion for the greater good
Human Analogy: Talking on a
Cell Phone
•  Alice and Bob talk on their cell phones
•  What if Bob couldn’t understand Alice?
–  ..or there was a brief dropout?
–  Bob asks Alice to repeat what she said
•  What if Bob hasn’t heard Alice for a while?
–  Is Alice just being quiet?
–  Or, have Bob and Alice lost connection?
–  Maybe Alice should periodically say “uh huh”
–  … or Bob should ask “Can you hear me now?” 
–  How long should Bob just keep on talking?
Highlights from Previous
Example
•  Acknowledgments from receiver
–  Positive: “okay” or “ACK”
–  Negative: “please repeat that” or “NACK”
•  Timeout by the sender (“stop and wait”)
–  Don’t wait indefinitely without receiving some
response
–  … whether a positive or a negative
acknowledgment
•  Retransmission by the sender
–  After receiving a “NACK” from the receiver
–  After receiving no feedback from the receiver
TCP Support for Reliable
Delivery
•  Checksum
–  Used to detect corrupted data at the receiver
–  …leading the receiver to drop the packet
•  Sequence numbers
–  Used to detect missing data
–  ... and for putting the data back in order
•  Retransmission
–  Sender retransmits lost or corrupted data
–  Timeout based on estimates of round-trip time
–  Fast retransmit algorithm for rapid retransmission
TCP Segments
TCP “Stream of Bytes”
Service
Byte 80
Byte 3
Byte 2
Byte 1
Byte 0
Byte 80
Byte 3
Byte 2
Byte 1
Byte 0
Host A
Host B
…Emulated Using TCP
Host A
“Segments”
Byte 0
Byte 1
Byte 2
Byte 3
Byte 80
Segment sent when:

TCP Data 1.  Segment full (Max Segment Size),
2.  Not full, but times out, or
3.  “Pushed” by application.
TCP Data
Host B
Byte 0
Byte 1
Byte 2
Byte 3
Byte 80
TCP Segment
IP Data
TCP Data (segment) TCP Hdr IP Hdr
•  IP packet
–  No bigger than Maximum Transmission Unit
(MTU)
–  E.g., up to 1500 bytes on an Ethernet
•  TCP packet
–  IP packet with a TCP header and data inside
–  TCP header is typically 20 bytes long
•  TCP segment
–  No more than Maximum Segment Size (MSS)
bytes
–  E.g., up to 1460 consecutive bytes from the
stream
Host A
Sequence Numbers
ISN (initial sequence number)
Sequence TCP
TCP Data
number = 1st HDR
byte ACK sequence

number = next
expected byte
TCP
TCP Data HDR
Host B
Initial Sequence Number (ISN)
•  Sequence number for the very first byte
–  Why not a de facto ISN of 0?
•  Practical issue
–  IP addresses and port #s uniquely identify a connection
–  Eventually, though, these port #s do get used again
–  … and there is a chance an old packet is still in flight
–  … and might be associated with the new connection
•  Security issue
–  An adversary can guess ISNs and hijack a connection
•  So, TCP requires changing the ISN over time
–  Set from a 32-bit clock that ticks every 4 microseconds
–  … which only wraps around once every 4.55 hours!
•  But, this means the hosts need to exchange ISNs
TCP Three-Way Handshake
Establishing a TCP Connection
A B
SYN
C K Each host tells

SYN A
its ISN to the
ACK other host.!
Data
Data
•  Three-way handshake to establish connection

–  Host A sends a SYN (open) to the host B
–  Host B returns a SYN acknowledgment (SYN ACK)
–  Host A sends an ACK to acknowledge the SYN ACK
TCP Header
Source port Destination port
Sequence number
Flags: SYN
Acknowledgment
FIN
RST HdrLen 0 Flags Advertised window
PSH
URG Checksum Urgent pointer
ACK Options (variable)
Data
Step 1: A’s Initial SYN Packet
A’s port B’s port
A’s Initial Sequence Number

Flags: SYN
Acknowledgment
FIN
RST 20 0 Flags Advertised window
PSH
A tells B it wants to open a connection…!

Step 2: B’s SYN-ACK Packet
B’s port A’s port
B’s Initial Sequence Number

Flags: SYN
A’s ISN plus 1
FIN
PSH
B tells A it accepts, and is ready to hear the next byte…!
… upon receiving this packet, A can start sending data!

Step 3: A’s ACK of the SYN-ACK
A’s port B’s port
Sequence number
Flags: SYN
B’s ISN plus 1
FIN
PSH
A tells B it wants is okay to start sending!
… upon receiving this packet, B can start sending data!

What if the SYN Packet Gets
Lost?
•  Suppose the SYN packet gets lost
–  Packet is lost inside the network, or
–  Server rejects the packet (e.g., listen queue is full)
•  Eventually, no SYN-ACK arrives
–  Sender sets a timer and wait for the SYN-ACK
–  … and retransmits the SYN-ACK if needed
•  How should the TCP sender set the timer?
–  Sender has no idea how far away the receiver is
–  Hard to guess a reasonable length of time to wait
–  Some TCPs use a default of 3 or 6 seconds
SYN Loss and Web Downloads
•  User clicks on a hypertext link
–  Browser creates a socket and does a “connect”
–  The “connect” triggers the OS to transmit a SYN
•  If the SYN is lost…
–  The 3-6 seconds of delay may be very long
–  The user may get impatient
–  … and click the hyperlink again, or click “reload”
•  User triggers an “abort” of the “connect”
–  Browser creates a new socket and does a “connect”
–  Essentially, forces a faster send of a new SYN packet!
–  Sometimes very effective, and the page comes fast
What’s Next
•  Read Chapter 1, 2, 3, 4.1-4.3, and 5.1-5.2
•  Next Lecture Topics from Chapter 5.3 and 5.4
–  UDP and TCP
•  Homework
–  Due Thursday in lecture
•  Project 3
–  Posted on the course webiste

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CS 457 – Lecture 20

Transport Layer: UDP and TCP

segments into messages, application

passes to application layer transport

• Lightweight communication between processes

SRC port DST port

“Address for www.cnn.com?”

Segment sent when:

byte ACK sequence

C K Each host tells

• Three-way handshake to establish connection

Source port Destination port

A’s Initial Sequence Number

A tells B it wants to open a connection…!

B’s Initial Sequence Number

B tells A it accepts, and is ready to hear the next byte…!

… upon receiving this packet, A can start sending data!

A tells B it wants is okay to start sending!

… upon receiving this packet, B can start sending data!

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•  Lightweight communication between processes

•  Three-way handshake to establish connection