The Transport Layer

1
The Transport Layer

• Chapter 6

2
Services Provided to the Upper Layers

• connection-oriented and connectionless
• very similar to the network layer
• transport layer makes it possible for the
  transport service to be more reliable than the
  underlying network service.
• Furthermore, the transport service primitives can
  be designed to be independent of the network
  service primitives, which may vary considerably
  from network to network.
• Dominant for network layer IP thus
  connectionless
• for transport layer
• TCP connection oriented
• UDP for connectionless IP with a little extra

3
Transport Service Primitives
4
Berkeley Sockets
SOCKET Create a new communication end point. The
parameters specify the addressing format, the
type of service (e.g. reliable byte stream) and
the protocol. It returns an ordinary file
descriptor for use in succeeding calls, (same as
OPEN). BIND Binds a local address to a
socket. LISTEN Allocates space to queue incoming
calls for the case that several clients try to
connect at the same time, it does not
block. ACCEPT Blocks a server until a connect
request TPDU from a client arrives. A new socket
with the same properties as the original LISTEN
one is then created by the transport entity and a
file descriptor for it is returned. The server
can then fork off a process or tread to handle
the connection on the new socket and go back to
waiting for the next connection on the original
socket by issuing a new ACCEPT primitive. CONNECT
Issued by a client after it has created a
socket, it blocks the client and actively starts
the connection process. When it completes (the
appropriate TPDU is received ), the client
process is unblocked and the connection is
established. SEND Sends some data over the
connection. RECEIVE Receives some data from the
connection CLOSE Symmetric connection release.
5
Transport vs Data link Protocol

• Both have to deal with error control, sequencing,
  flow control and other issues.
• In the subnet a packet might be stored for a
  number of seconds and then delivered later. This
  can sometimes be disastrous, specially during
  connection and disconnection, and requires the
  use of special protocols.
• The large and dynamically varying number of
  connections in the transport layer, requires a
  different buffering and flow control approach
  than in the data link layer. The idea of
  dedicating many buffers to each connection in the
  transport layer is not attractive.

6
Flow control

• In data link and transport layer a sliding window
  or other scheme is needed on each connection to
  keep a fast transmitter from overrunning a slow
  receiver.
• A router has relatively few lines whereas a host
  may have many open connections.
• The optimum tradeoff between source and
  destination buffering depends on the reliability
  of the connection and the type of traffic.
• For low bandwidth, bursty traffic (like produced
  by an interactive terminal) it is better to
  buffer at the sender. For high bandwidth, smooth
  traffic (like a file transfer) it is better to
  buffer at the receiver.
• Dynamically adjustment of buffer allocation is
  needed.

7
Decouple buffering

• A general way to manage dynamic buffer allocation
  is to decouple the buffering from the
  acknowledgments, in contrast to the sliding
  window protocol of the data link layer.
• Initially the sender requests a certain amount of
  buffer space on the receiver side, based on its
  perceived needs. The receiver grants as many as
  it can afford.
• Every time the sender transmits data, it must
  decrement its allocation, stopping all together
  if it reaches 0.
• The receiver separately piggybacks both acks and
  the amount of available buffer space onto the
  reversed traffic.

8
UDP

• UDP (User Datagram Protocol) is just basically IP
  with a short header added. The port numbers
  indicate the sending and receiving transport end
  points. When a UDP packet arrives, its payload is
  send to the process attached to the destination
  port.
• The checksum is optional and stored as 0 if not
  computed, a calculated 0 checksum is stored as
  all 1s. UDP does not do flow control, error
  control or retransmission upon receipt of a bad
  datagram. All of that is up to the user process.

9
Internet Transport Protocol TCP

• TCP (Transmission Control Protocol) provides a
  reliable byte stream over an unreliable
  internetwork.
• TCP accepts user data streams from local
  processes, breaks them up into pieces not
  exceeding 64K (usually about 1460 bytes to fit in
  a single Ethernet frame) and sends each piece as
  a separate IP datagram.
• The receiver side gives IP datagrams containing
  TCP data to its TCP entity, which reconstructs
  the original byte streams.
• IP gives no guarantees that datagrams will be
  delivered properly, so it is up to TCP to time
  out and retransmit. Also IP datagrams might be
  delivered in the wrong order, it is up to TCP to
  rearrange them in the proper sequence.

10
The TCP Service Model
Port
Protocol
Use
Port numbers below 1024 are called well-known
ports and are reserved for standard services.
21
FTP
File transfer
23
Remote login
Telnet
E-mail
25
SMTP
69
Trivial File Transfer Protocol
TFTP
Finger
Lookup info about a user
79
80
World Wide Web
HTTP
POP-3
110
Remote e-mail access
USENET news
119
NNTP
The TCP service is obtained by having both the
sender and receiver create end points, called
sockets. Each socket has a socket number
(address) consisting of the IP address and a 16
bit number local to that host, called a port. Two
or more connections may terminate at the same
socket. Connections are identified by the socket
identifiers at both ends, that is (socket1,
socket2). All TCP connections are full-duplex
(traffic can go in both directions) and
point-to-point (each connection has exactly 2 end
points). Multicasting or broadcasting are not
supported.
11
Byte stream
A TCP connection is a byte stream, not a message
stream. For example, if the sending process does
four 512 byte writes to a TCP stream, these data
may be delivered to the receiving process as four
512 byte chunks, two 1024 bytes chunks, one 2048
byte chunk or some other way. This is analogous
to UNIX files. TCP may send user data
immediately or buffer it, in order to send larger
segments to the network layer (larger IP
datagrams.) Applications might use the PUSH flag
to indicate that TCP should send the data
immediately. The application can also send urgent
data (e.g. when an interactive user hits the
CTRL-C key), TCP then puts control information in
its header and the receiving side interrupts
(gives a signal in UNIX terms) to the application
program using the connection.
12
TCP Segment Header
13
The TCP Checksum

• It checksums the header, the data and a
  conceptual pseudo header.
• The pseudo header contains IP addresses and thus
  violates the protocol hierarchy. It helps to
  detect wrongly delivered packets.
• The checksum is a simple one, just adding the
  16-bits words in 1's complement and then take the
  1's complement of the sum. The receiving side
  should find a 0.

14
TCP Connection Establishment
A 3-way handshake. CR (connection request) and
REJECT are indicated by the SYN and FIN bits in a
segment header. The initial sequence numbers are
chosen in a special way, ensuring that no delayed
or duplicated older request can fake a connection.
15
TCP Connection Management Modeling
There are 11 states used in the TCP connection
management finite state machine. Data can be send
in the ESTABLISHED and the CLOSE WAIT states and
received in the ESTABLISHED and FINWAIT1 states.
Releasing a TCP connection is symmetric. Either
part can send a TCP segment with the FIN bit set,
meaning it has no more data to send. When the FIN
is acked, that direction is shut down, but data
may continue to flow in the other direction. It
is not really fool proof, but it works.
16
TCP Finite state Machine

• Each line is marked by an event/action pair.
• The event can either be a user-initiated system
  call (CONNECT, LISTEN, SEND or CLOSE), a segment
  arrival (SYN, FIN, ACK or RST), or a timeout.
• For the TIMED WAIT state the event can only be a
  timeout of twice the maximum packet length.
• The action is the sending of a control segment
  (SYN, FIN or RST) or nothing.
• The time-outs to guard for lost packets ( e.g. in
  the SYN SENT state) are not shown here.

17
TCP Window management
Sender buffering Senders are not required to
transmit data as soon as they come in from the
application. Usually Nagle's algorithm is used.
When data come into the sender one byte at a time
(e.g. on a Telnet connection), just the first
byte is send and the rest is buffered until the
outstanding byte is acked. Sometimes it is better
to disable this, e.g. when mouse movements are
sent in X-Windows applications.
18
Silly window syndrome

• This occurs when the sender transmit data in
  large blocks, but an interactive receiver
  application reads data 1 byte at a time.
• The receiver continuously gives 1 byte window
  updates and the sender transmits 1 byte segments.
• Clark's solution is to let the receiver only send
  updates when it can handle the maximum segment
  size it advertised when the connection was
  established or when its buffer is half empty.
• Receivers are also not required to send acks and
  window updates immediately.
• Many implementations delay them for 500 msec in
  the hope to piggyback them.

19
TCP Congestion Control

• It is assumed that time-outs are caused by
  network or receiver congestion as lost packets
  are rare these days.(Wireless networks are now
  an exception)
• Each sender maintains two windows
• the window the receiver has granted (indicating
  the receiver buffer capacity)
• the congestion window (indicating the network
  capacity).
• The number of bytes that may be sent is the
  minimum of the 2 windows.

20
TCP slow start
The congestion window is doubled on each packet
successfully sent (an ack received before the
timeout). This exponential increase (called the
slow start) continues until the threshold
(initially 32K) is reached, after which the
increase is linearly.
When a timeout occurs, the threshold is set to
half the current congestion window and the slow
start is repeated. If an ICMP source quench
packet comes in, it is treated the same way as a
timeout.
21
TCP Timer Management
For each segment the round trip time M is
measured and the estimates of the mean and mean
deviation are updated as    RTT ßRTT
(1-ß)M    D ßD (1-ß) RTT-Mwith ß a
smoothing parameter, typically 7/8. The timeout
is then set to RTT 4 D
22
Improvements

• Change from the original Go-Back-N to an more
  Selective Repeat type of retransmit of lost
  segments
• Buffering out-of-order segments in the receiver
• Sender detects missing segments by looking at
  duplicate ACKs (NACKs can not be used) and
  retransmits before the timer expires.
• Improved congestion control
• look also to duplicate ACKs to lower the
  congestion window to half its value
• start from that value with linear increase
• versions to handle HTTP (www) trafic efficiently
• versions for high speed links to increase the
  throughput
• On a wireless network if a packet is lost, it is
  usually not due to congestion, but due to
  transmission "noise". TCP should not slow down,
  but retransmit as soon as possible.

23
Wireless TCP

• Using an indirect TCP solution is a possibility.
  But the acknowledgment, returned from the base
  station to the sender, does not indicate that the
  mobile host has received the data.
• Another possibility is to add a snooping agent on
  the base station. It watches the interchange
  between base station and mobile host and performs
  retransmissions (and interception of duplicated
  acknowledgments) on that part alone. However
  there is still the possibility that the sender
  times out and starts it congestion control.

24
Real-Time Transport Protocol

• RTP is intended for real-time multimedia
  applications
• Its basic function is to multiplex several
  real-time data streams into a single stream of
  UDP packets
• It can be regarded as a transport sublayer

• Used for e.g. radio, telephony, music-on-demand,
  videoconferencing, video-on-command, etc.
• It may contain for example a video stream and 2
  audio stream for stereo or sound tracks in 2
  languages.

25
RTP header

• Each payload may contain multiple samples, they
  may be coded any way the user wants. The payload
  type field indicates which one is used.
• The timestamp gives the time of each sample
  relative to the first sample.
• Synchronization source identifier uniquely
  identifies the synchronizing source of a stream
• Contributing source identifier uniquely
  identifies the contributing streams (There are CC
  of them)

26
some RTP audio/ video formats
27
Real Time Control Protocol

• RTCP can be used to handle synchronization
  between the sources.
• further it gives feedback on delay, jitter,
  bandwidth and congestion.
• The encoding process can use it to increase the
  data rate and give better quality if the network
  is functioning well,
• otherwise it can cutback in data rate and hence
  reduce the quality.
• note that missing RTP packets are usually not
  requested but are to some extend compensated for
  in the used encoding method.

28
Real-Time Streaming Protocol
RTSP controls the transmission of a video
stream. It does not control the stream itself.