Faculty:

21CS52 – COMPUTER NETWORKS Prof. Ashok Herur

[Link]@[Link]
Mobile: 91641 01399
 Module 4

Transport Layer

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 Main topics in Module 4

• The Transport service

• Different transport layer protocols – TCP and UDP

• Flow control and Error control

• Congestion control

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OSI Reference model

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 Functions of the Transport Layer

• It is the lowest host-to-host protocol.

• Facilitates communication between host processes.

• Divides the message into data segments on the sending side, and reassembles
them on the receiving side.

• Takes complete responsibility for a reliable, end-to-end transport of the

message.

• Packets can be: Corrupted, or Lost, or Delivered out of sequence.

• Also takes care of Flow control between the hosts.

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Segments in the Transport Layer

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 Transport entity

• The software and hardware within the Transport layer that does the work is
called the Transport entity.

• The Transport entity is generally located (when it comes to Internet) in the

Operating System, or in a library package bound into network applications.

• It can also be in a separate user process or even in the Network Interface card.

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 Transport layer addressing

• When an application process wishes to set up a connection to a remote

application process (on another host), it must specify it unambiguously.

• In the Internet, such addresses are called Ports. (in generic terms, an address
is called TSAP – Transport Service Access Point).

• This is similar to an IP address in the Network layer (NSAP – Network Service

Access Point).

• However, note that a host could be running many applications simultaneously,

in which case a single IP address will lead to many Port addresses.

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Transport layer addressing

The Transport layer at the

destination host should
know the port to which
the data is to be delivered.

Eg. shown is from TSAP

1208 on Host 1 to TSAP
1522 on Host 2.

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 Connection-oriented vs. Connectionless sessions

• The Network layer too had Connectionless (Datagram Packet Switching) and
Connection-oriented (Virtual Circuit Packet Switching) transmissions.
• However, they were restricted to transmission from the transmitting router
(representing the transmitting host) to the terminal router (representing the
destination host).
• Any deficiencies in the service has to be handled by the Transport layer.

• Transport layer has two main protocols:

• TCP (Transport Control Protocol) – Connection-oriented
• UDP (User Datagram Protocol) – Connectionless.

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 Transmission Control Protocol (TCP)

• Provides a reliable, end-to-end, byte-stream transmission over an unreliable

internetwork.

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 Transmission Control Protocol (TCP)

• Is Connection-oriented
• Explicit set-up and tear-down of TCP session
• Offers Stream-of-bytes service
• Receives a stream of bytes from higher layers.
• Passes them as Segments to the lower layers.
• Provides Reliable, in-order delivery
• Checksums to detect corrupted data
• Acknowledgments & retransmissions for reliable delivery
• Sequence numbers to detect losses and reorder data

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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

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 TCP Header fields
• Port Number:
• A port number identifies the endpoint of a connection.
• A pair <IP address, port number> identifies one endpoint of a connection.

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 TCP Header fields

• Sequence Number:
• The range of the 32-bit SeqNo is
0 <= SeqNo <= 2 32 -1 ≈ 4.3 Giga
• Each sequence number identifies a byte in the byte stream.
• Initial Sequence Number (ISN) of a connection is set during connection
establishment.

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TCP Header fields

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 TCP Header fields

• Acknowledgement Number:
• Acknowledgements are piggybacked.
• A segment from A -> B can contain an acknowledgement for a data sent in
the B -> A direction.
• If a host sends an AckNo in a segment it sets the “ACK flag”
• The AckNo contains the SeqNo of the next byte that a host wants to
receive.
• Example: The acknowledgement for a segment with sequence numbers
0-1500 is AckNo=1501.

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 TCP Header fields

• Header Length:
• Length of header in terms of 32-bit words (4 bytes)
• The TCP header has a variable length (with minimum 20 bytes)
• This 4-bit field can hold numbers from 5 (20 bytes) to 15 (60 bytes)
• Flags:
• SYN: Synchronize sequence numbers
• Sent in the first packet when initiating a connection
• FIN: Sender is finished with sending
• Used for closing a connection
• Both sides of a connection must send a FIN

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 TCP Header fields
• Flags: (contd)
• RST: Causes the receiver to Reset the connection
• Receiver terminates the connection and indicates higher layer
application about the reset
• ACK: Acknowledgement Number is valid
• URG: Urgent pointer is valid
• If the bit is set, the following bytes, in the range given below,
contain an urgent message:
SeqNo <= urgent message <= SeqNo+urgent pointer
• PSH: PUSH flag is used to tell the receiver to pass all data (that it currently
has) to the application.
• Normally set by sender when the sender’s buffer is empty.
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TCP Header fields
• Window Size:
• Each side advertises the size of its sliding window.
• Window size is the maximum number of bytes that a receiver can accept.
• Maximum window size is 216-1= 65535 bytes
• TCP Checksum:
• TCP checksum covers over both TCP header and TCP data
• Urgent Pointer:
• Only valid if URG flag is set

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Establishing a connection (Hand-shaking)

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Establishing a TCP Connection: 3-way handshake.
A B
SYN

CK Each host tells its ISN (Initial

SYN A
sequence number) to the other host.
ACK
Dat
aD
at
a

• 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
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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
URG Checksum Urgent pointer
ACK
Options (variable)

A tells B it wants to open a connection…

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Step 2: B’s response (SYN-ACK) packet to A

B’s port A’s port

B’s Initial Sequence Number

Flags: SYN
A’s ISN plus 1
FIN
RST 20 0 Flags Advertised window
PSH
URG Checksum Urgent pointer
ACK
Options (variable)

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

… upon receiving this packet, A can start sending data

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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
RST 20 0 Flags Advertised window
PSH
URG Checksum Urgent pointer
ACK
Options (variable)

A tells B it is now okay to start sending it’s own data (to

A)
… upon receiving this packet, B can start sending
data

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Tearing down a TCP Connection

A FI
B
N
CK
FIN A
At this time, B can still send
data to A
FIN
FIN
ACK

• Each end of the data flow must be shut down independently.

• If one end is done, it sends a FIN segment. This means that no
more data will be sent.
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Error control in TCP

• The Data Link layer implements error control measures like using
a CRC Checksum and using some ARQ protocols.
• If every frame is verified, on EVERY link, at the Data Link layer, can
something be wrong, end-to-end, when the message reaches the
destination host?
• While the Link layer checks take care of the errors that may occur
on the various links, they cannot take care of the errors that may
occur INSIDE the routers.
• At every router, various fields (TTL, Fragment Offset, etc) will /
may change, leading to a change in the Packet Header checksum.

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Flow control in TCP

• The Data Link layer implements flow control measures like using
Sliding window protocols.
• What would be the optimum size of a window?
• At the Data Link layer (for a link between adjacent nodes), the
window size can be pretty small (1, 2, 4,…) since the Bandwidth-
Delay product is low.
• Bandwidth-Delay product broadly indicates how many bits
(segments) could have been transmitted by the time a sender
receives the acknowledgment of a message segment sent earlier.

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Flow control in TCP

• However, at the Transport Link layer (host to host), the

Bandwidth-Delay product will be high because of the high delay
required to traverse the entire path to the destination.
• Therefore, the Window size here (at the Transport Layer) has to
be much larger.
• The same logic applies to the Buffering at the transmitting host.
• It has to be much larger than that used at an intermediate
router.

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Wireless issues in TCP

• Transport layer protocols such as TCP that implement congestion

control should be independent of the underlying Network and
Data Link layer technologies.
• But, in practice, there are various issues with wireless links /
networks.
• The main issue is that % packet loss is often used as a measure of
congestion.
• This % packet loss is normally high on wireless links / networks,
due to the higher rate of transmission error (due to noise).

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Wireless issues in TCP

• By the time, this % packet loss reaches around 10%, the

connection is effectively stopped because of the throttling by the
congestion control mechanisms.
• One solution is to differentiate the packet losses due to
transmission error and due to insufficient bandwidth.
• Things will be more complicated when the path contains a few
wired links and a few wireless links.

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User Datagram Protocol (UDP)

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User Datagram Protocol (UDP)
• The main characteristics of UDP are as follows
• UDP implements a connectionless transfer.
• It does not require connection establishment prior to data transfer
• UDP service is unreliable
• UDP does not guarantee the delivery of datagram to the destination
• It does not guarantee the correct order of the datagrams.
• UDP computes the checksum for the entire header plus data
• UDP does not buffer the incoming datagrams.
• UDP does not have the feature to segment long messages.

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User Datagram Protocol (UDP)
• UDP results in speedier communication because it does not spend time
forming a firm connection with the destination before transferring the data.
• User Datagram Protocol (UDP) is used for:
• Sending relatively small amounts of data, eliminating concerns regarding
controlling errors or the flow of the packets
• Routing update protocols such as Routing Information Protocol (RIP) and
Network control protocols like SNMP.
• Multicasting because UDP works well with packet switching.

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User Datagram Protocol (UDP)
• Real-time applications in which the information needs to be delivered quickly
and smoothly.
• It is specifically chosen for time-sensitive applications like gaming, playing
videos, or Domain Name System (DNS) lookups (convert Domain names to IP
addresses).
• For users, it is better to have the overall transmission arrive on time than wait
for it to get there in a near-perfect state.
• For this reason, UDP is commonly used in Voice over Internet Protocol
(VoIP) applications as well.

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UDP header

0 15, 16
31

Source Port Number Destination Port Number

UDP Length Checksum

Data

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User Datagram Protocol (UDP)
• Source port number identifies the sender's port, when used, and should be
assumed to be the port to reply to if needed.
• If not used, it should be zero.
• If the source host is the server, the port number is likely to be a well-known
port number from 0 to 1023.
• If the source host is the client, the port number is likely to be an ephemeral
(temporary, short-lived) port number.
• In networking, you might use well-known ports (0 to 1023), registered ports
(1024 to 49151), and dynamic or ephemeral ports (49152 to 65535), as
defined by the Internet Assigned Numbers Authority (IANA).
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User Datagram Protocol (UDP)
• Destination port number identifies the receiver's port and is required.
• UDP Length field specifies the length in bytes of the UDP header and data.
• The minimum length is 8 bytes, the length of the header.
• The field size sets a theoretical limit of 65,535 bytes (8-byte header +
65,527 bytes of data) for a UDP datagram.
• The Checksum field may be used for error-checking of the header and data (by
the Application layer); The field carries all-zeros if unused.

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Congestion control

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Congestion control
• Congestion control has to ensure that the subnet is able to carry the traffic
offered by the hosts.
• It is a global issue, involving all hosts, all routers and all links in the
network.
• When too many packets are present in a small part of the subnet, it leads to
congestion.
• Then, the data carrying capacity of the subnet reduces.
• When the number of packets dumped onto the subnet by the hosts is within
the carrying capacity, they are all delivered (except the ones with
transmission errors)
• The number delivered is proportional to the number sent by the hosts
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Congestion control
• As traffic increases beyond the capacity, the network cannot cope and
packets are lost.

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Causes of congestion
• Load greater than the resources in a part of the network
• More traffic and low bandwidth lines
• Insufficient buffer at the routers
• Having a very large buffer is counter-productive (why?)
• Slow processors at the routers

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Measures of congestion
• Percentage of packets discarded for lack of buffer space
• Average queue length at the routers
• Average packet delay
• Standard deviation of the packet delay
• Number of packets that are timed-out and retransmitted
In all the above cases, a rising number indicates growing congestion.

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Congestion control algorithms
• In spite of the routers adapting to the traffic conditions, and changes in
topology (adaptive routing), traffic can build up in a part of the network.
• The basic principle behind all the Congestion Control algorithms is: Reduce
load and / or Increase capacity.
• Often, it is difficult to increase capacity temporarily and at short notice.
• Therefore, most of the algorithms concentrate on reducing (regulating)
the load.

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Congestion control algorithms
• Common techniques :
• Admission Control techniques:
• Leaky bucket algorithm

• Token bucket algorithm

• Traffic throttling techniques:

• Choke packet algorithm

• Explicit Congestion Notification

• Load shedding

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Leaky Bucket algorithm

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Leaky Bucket algorithm
• The leaky bucket tries to ensure that the number of packets put out on to the
network (on an average) is constant, regardless of the burstiness of the input;
It does nothing when input is idle or the rate is below the defined constant
rate.
• The host injects one packet per clock tick onto the network. This results in a
uniform flow of packets, smoothing out bursts and reducing congestion.
• It is implemented as a single-server queue with constant service time.
• If the bucket (buffer) overflows, then packets are discarded.

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Leaky Bucket algorithm
• When packets are of variable length, it is better to allow a fixed number of
bytes per tick.
• The leaky bucket algorithm imposes the same restriction on all hosts (some
sending almost continuously and some sending occasionally).
• In short, it does not allow any burstiness beyond the constant output rate
that has been defined.

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Token Bucket algorithm
• The Token Bucket algorithm allows the output rate to vary, depending on the
size of the burst.
• In the TB algorithm, the bucket holds tokens.
• Each token entitles the host to transmit a fixed number of bytes.
• To transmit a packet, the host must capture and destroy the required
number of tokens (depending on the packet size).
• Tokens are generated by a clock at the rate of one token every Δt sec.
• Idle hosts can capture and save up tokens (up to the max. size of the bucket)
in order to send larger bursts later.
• When the bucket fills up, new tokens are lost.

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Token Bucket algorithm

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Choke Packet algorithm
• When a router is congested or about to get congested, it starts sending choke
packets to the sources from where it received packets.
• A choke packet is a control packet generated at a congested node.
• The source, on receiving the choke packet must reduce its transmission rate
by a certain percentage.
• This is a direct way of telling the sources to slow down temporarily.

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Choke Packet algorithm

D is the congested router, and A is the source of the congestion

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Hop-by-hop Choke Packet algorithm
• Over long distances or at high speeds, choke packets are not very effective.
• A more efficient method is to send to choke packets hop-by-hop.
• This requires each hop to reduce its transmission even before the choke
packet arrive at the source.

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Hop-by-hop Choke Packet algorithm

• This technique will spread the

congestion backwards – To the routers
along the way.

• It is better to have many routers ‘mildly’

congested than having one router
congested ‘heavily’

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Explicit Congestion Notification (ECN)
• Instead of generating additional (choke) packets to warn of congestion, a
router can tag any packet it forwards (by setting a bit in the packet’s header)
to signal that it is experiencing congestion.
• When the packet is delivered, the destination notes that there is congestion
and informs the sender when it sends a Reply / Acknowledgement packet.
• The sender can then throttle its transmissions, as before.
• This method is called Explicit Congestion Notification (ECN).

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Load shedding
• This method is the last resort, when none of the milder methods, described
earlier, are able to control congestion.
• Load shedding just means that the congested router will throw away packets
to reduce the load, when the buffer is filled up / or about to get filled up.
• The key question here is about which packets to discard; The answer depends
on the type of application to which the packets belong.
• For a File Transfer, an old packet is worth more than a new one. Dropping packet 6 and
keeping packets 7 to 10 (for example) will only force the receiver to do more work to
buffer the data that it cannot yet use.
• In contrast, in real-time applications, a new packet is worth more than an old one,
because packets become useless if they are delayed and miss the time when they must
be played out.

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Load shedding – Random Early Detection (R.E.D.)
• Dealing with congestion when it first starts is more effective than dealing
with it when it gets worse.
• Random Early Detection is a type of Load Shedding that discards packets
before the buffer space is filled up.
• This idea stems from the fact that most hosts on the Internet do not get
congestion signals in the form of ECN.
• The only reliable indication of congestion that they get is Packet loss.
• To determine when to start discarding, routers maintain a running average of
their queue length.

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Load shedding – Random Early Detection (R.E.D.)
• When the average queue length on some link exceeds a threshold, that
particular link is said to be congested, and a small fraction of the packets
(coming on that link) are dropped at random.
• Picking packets at random makes it more likely that the fastest senders will
see more packet drops.
• This is the best option since a router cannot tell which of the sources
(sending packets on that link) is contributing more to the congestion.
• The affected sender will notice the loss, and its Transport layer protocol will
slow down.
• The (higher-than-normal) lost packets are thus delivering the same
message as a choke packet, but implicitly.

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