UNIT-IV: TRANSPORT LAYER

UDP
1. Segment header
2. Remote Procedure Call (RPC).
3. Real-time Transport Protocols (RTP).
TCP
1. service model.
2. Protocol.
3. Segment header.
4. Connection establishment .
5. Connection release .
6. Sliding window .
7. Timer management.
8. Congestion control. 1
 The Internet Transport Protocol: UDP
• Introduction to UDP (User Datagram Protocol).
• Segment header
Two applications of UDP:
1. Remote Procedure Call (RPC).
2. The Real-Time Transport Protocol (RTP).

2
 Introduction to UDP
• The Internet has two main protocols in the Transport
Layer, a Connectionless Protocol and Connection-
oriented Protocol.
• UDP is a transport layer protocol that typically runs in
the operating system and protocols that use UDP
typically run in user space.

The UDP Header.

3
• The packet produced by the UDP is called a user
datagram.
– Source port address: The source port address is the
address of the application program that has created
the message.
– Destination port address: The destination port
address is the address of the application program that
will receive the message.
– Total length: The total length field defines the total
length of the user datagram in bytes.
– Check sum: The check sum is a 16-bit field used in
error detection.
4
The IPv4 pseudoheader included in the UDP checksum

• Including the pseudoheader in the UDP checksum

computation helps detect misdelivered packets.
• UDP can discover that an error has occurred; ICMP can
then inform the sender that a user datagram has been
damaged and discarded.
5
• It is probably worth mentioning explicitly some of the
things that UDP does not do.
– It does not do Flow Control, Error Control, or
Retransmission upon receipt of a Bad Segment.
– All of that is up to the User Processes.
• What it does do is provide an interface to the IP
protocol with the added feature of demultiplexing
multiple processes using the ports.
• One area where UDP is especially useful is in client-
server situations. Often, the client sends a short request
to the server and expects a short reply back.

6
 Applications of UDP
• Client- Server Situations: The client sends a short
request to the server and expects a short reply back. If
either the request or the reply is lost, the client can
just time out and try again.
• DNS (Domain Name System): for example,
www.cs.berkeley.edu, can send a UDP packet
containing the host name to a DNS server. The server
replies with a UDP packet containing the host’s IP
address. No setup is needed in advance and no release
is needed afterward. Just two messages go over the
network.

7
UDP - Client Server Application

8
 Transmission Control Protocol (TCP) User Datagram Protocol (UDP )
TCP supports host-to-host UDP enables process-to-process
communication I,e. Connection oriented communication, I,e. Connectionless protocol
protocol
The most widely used protocol in the Used for voice over IP, streaming video,
internet Gaming and Live Broadcasting
No packet loss during transmission Faster and need Few Resources
Send packets in order Packets don’t necessarily arrive in order
Slower and requires more resources Allows missing packets, but the sender is
unable to know whether a packet has been
received.
Best suitable for high reliability and Best suitable for applications that need fast,
transmission time is relatively less critical efficient transmission, such as games.
TCP sends individual packets and is UDP sends messages, called datagrams and
considered a reliable transport medium. is considered a best-effort mode of
communications.
TCP provides error and flow control, no UDP is considered a connectionless protocol
such mechanisms are supported in UDP. because it doesn't require a virtual circuit to
be established before any data transfer
occurs 9
 Remote Procedure Call
• In a certain sense, sending a message to a remote host
and getting a reply back is a lot like making a function
call in a programming language.
• Birrell and Nelson suggested was allowing programs to
call procedures located on remote hosts.
• When a process on machine 1 calls a procedure on
machine 2, the calling process on 1 is suspended and
execution of the called procedure takes place on 2.
• Information can be transported from the caller to the
callee in the parameters and can come back in the
procedure result.
• No message passing is visible to the programmer. This
technique is known as RPC (Remote Procedure Call) and
has become the basis for many networking applications.10
• In the simplest form, to call a remote procedure, the
client program must be bound with a small library
procedure, called the client stub, that represents the
server procedure in the client's address space.
• Similarly, the server is bound with a procedure called
the server stub.

The actual steps in making an RPC 11

Steps in making a Remote Procedure Call
Step 1 is the client calling the client stub. This call is a local procedure call, with
the parameters pushed onto the stack in the normal way.
Step 2 is the client stub packing the parameters into a message and making a
system call to send the message. Packing the parameters is called
marshaling (assembling).
Step 3 is the kernel sending the message from the client machine to the server
machine.
Step 4 is the kernel passing the incoming packet to the server stub.
Step 5 is the server stub calling the server procedure with the unmarshaled
parameters. The reply traces the same path in the other direction. 12
 The Real-Time Transport Protocol
• Client-server RPC is one area in which UDP is widely
used.
• Another one is Real-time Multimedia Applications, as
Internet radio, Internet telephony, music-on-
demand, videoconferencing, video-on-demand, and
other multimedia applications.
• Thus was RTP (Real-time Transport Protocol) born.
• RTP normally runs in user space over UDP (in the
operating system).

13
 The Real-Time Transport Protocol

(a) The position of RTP in the protocol stack.

(b) Packet nesting.
The two aspects of Real-time Transport Protocol: The first is the RTP
protocol for transporting audio and video data in packets.
The second is the processing that takes place, mostly at the receiver,
to play out the audio and video at the right time.
14
• As a consequence of this design, it is a little hard to say which layer
RTP is in.
RTP—The Real-time Transport Protocol
• The basic function of RTP is to multiplex several real-time data
streams onto a single stream of UDP packets.
• The UDP stream can be sent to a single destination (unicasting) or
to multiple destinations (multicasting).
• Because RTP just uses normal UDP, its packets are not treated
specially by the routers unless some normal IP quality-of-service
features are enabled.
• In particular, there are no special guarantees about delivery, jitter,
etc.
• Jitter is the variation in time delay between when a signal is
transmitted and when it's received over a network connection.
This is often caused by network congestion, poor hardware
performance and not implementing packet prioritization.
15
• Each packet sent in an RTP stream is given a number
one higher than its predecessor.
• This numbering allows the destination to determine if
any packets are missing.
• If a packet is missing, the best action for the destination
to take is to approximate the missing value by
interpolation.
• Retransmission is not a practical option since the
retransmitted packet would probably arrive too late to
be useful.
• As a consequence, RTP has no flow control, no error
control, no acknowledgements, and no mechanism to
request retransmissions.

16
The Real-Time Transport Protocol Header

Ver- Version
P- Packet
X- Extension Header
CC- Contributing Sources
M- Application-specific Mark
17
• It consists of three 32-bit words and potentially some
extensions.
• The first word contains the Version field, which is
already at 2.
• The P bit indicates that the packet has been padded
to a multiple of 4 bytes.
• The last padding byte tells how many bytes were
added.
• The X bit indicates that an Extension Header is
present.
• The format and meaning of the extension header are
not defined. The only thing that is defined is that the
first word of the extension gives the length.

18
• The CC field tells how many Contributing Sources are
present. It can be used to , from 0 to 15 .
• The M bit is an Application-specific Mark the start of
a video frame, the start of a word in an audio
channel, or something else that the application
understands.
• The Payload type field tells which encoding algorithm
has been used (e.g., uncompressed 8-bit audio, MP3,
etc.).
• The Sequence number is just a counter that is
incremented on each RTP packet sent. It is used to
detect lost packets.

19
• The Timestamp is produced by the stream's source
to note when the first sample in the packet was
made. This value can help reduce jitter.
• The Synchronization source identifier tells which
stream the packet belongs to. It is the method used
to multiplex and demultiplex multiple data streams
onto a single stream of UDP packets.
• The Contributing source identifiers, if any, are used
when mixers are present in the studio. In that case,
the mixer is the synchronizing source, and the
streams being mixed are listed here.

20
The Internet Transport Protocols: TCP
Introduction to TCP (Transmission Control Protocol)
1. The TCP Service Model
2. The TCP Protocol
3. The TCP Segment Header
4. TCP Connection Establishment
5. TCP Connection Release
6. Sliding Window
7. TCP Timer Management
8. TCP Congestion Control
21
 Introduction to TCP
(Transmission Control Protocol)
• UDP is a simple protocol and it has some very important uses, such
as client server interactions and multimedia, but for most Internet
applications, reliable, sequenced delivery is needed.
• UDP cannot provide this, so another protocol is required. It is
called TCP and is the main workhorse of the Internet.
• TCP (Transmission Control Protocol) was specifically designed to
provide a reliable end-to-end byte stream over an unreliable
internetwork.
• TCP was designed to dynamically adapt to properties of the
internetwork and to be robust in the face of many kinds of
failures.

22
 1. The TCP Service Model
• TCP service is obtained by both the sender and the receiver
creating end points, called sockets.
• Each socket has a socket number (address) consisting of the IP
address of the host and a 16-bit number local to that host,
called a port.
• A port is the TCP name for a TSAP : A Transport Services Access
Point is an end-point for communication between
the Transport layer and the Session layer in the OSI (Open
Systems Interconnection) reference model.
• Each TSAP is an address that uniquely identifies a
specific instantiation of a service.
• TSAPs are created by concatenating the node's Network Service
Access Point (NSAP) with a transport identifier, and sometimes
a packet and/or protocol type. 23
A socket may be used for multiple connections at the
same time.
Connections are identified by the socket identifiers at
both ends, that is, (socket1, socket2). No virtual circuit
numbers or other identifiers are used.

The socket primitives for TCP 24

• Port numbers below 1024 are reserved for standard
services that can usually only be started by privileged
users (e.g., root in UNIX systems).
• They are called well-known ports.

25
• Secure Shell (SSH) is a cryptographic network protocol for
operating network services securely over an unsecured
network
• IMAP (Internet Message Access Protocol) is a standard email
protocol that stores email messages on a mail server, but
allows the end user to view and manipulate the messages as
though they were stored locally on the end user's computing
device(s).
• The Real Time Streaming Protocol (RTSP) is a network
control protocol designed for use in entertainment and
communications systems to control streaming media servers.
The protocol is used for establishing and controlling media
sessions between end points.
• The Internet Printing Protocol (IPP) is a specialized Internet
protocol for communication between client devices
(computers, mobile phones, tablets, etc.) and printers 26
All TCP connections are full duplex and point-to-point.
• Full duplex means that traffic can go in both directions at the
same time.
• Point-to-point means that each connection has exactly two end
points.
• TCP does not support Multicasting or Broadcasting.
• A TCP connection is a byte stream, not a message stream.
• Message boundaries are not preserved end to end.
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-byte chunks, one 2048-byte
chunk , or some other way.

27
 The TCP Service Model

(a) Four 512-byte segments sent as separate IP datagrams.

(b) The 2048 bytes of data delivered to the application in a
single READ CALL.

28
 2. The TCP Protocol
• A key feature of TCP, is that every byte on a TCP connection has its
own 32-bit sequence number.
• The sending and receiving TCP entities exchange data in the form
of segments.
• A TCP segment consists of a fixed 20-byte header (plus an
optional part) followed by zero or more data bytes.
• The TCP software decides how big segments should be. It can
accumulate data from several writes into one segment or can split
data from one write over multiple segments.
• Two Limits Restrict the Segment Size. First, each segment,
including the TCP header, must fit in the 65,515- byte IP payload.
Second, each link has an MTU (Maximum Transfer Unit).
• Each segment must fit in the MTU at the sender and receiver so
that it can be sent and received in a single, unfragmented packet.
In practice, the MTU is generally 1500 bytes (the Ethernet
payload size) and thus defines the upper bound on segment size.29
• The basic protocol used by TCP entities is the sliding window
protocol with a dynamic window size. When a sender transmits a
segment, it also starts a timer.
• When the segment arrives at the destination, the receiving TCP
entity sends back a segment bearing an acknowledgement
number equal to the next sequence number it expects to receive
and the remaining window size.
• If the sender’s timer goes off before the acknowledgement is
received, the sender transmits the segment again.

30
The advantages of TCP/IP protocol suite are
• It is an industry–standard model that can be effectively
deployed in practical networking problems.
• It is interoperable, i.e., it allows cross-platform communications
among heterogeneous networks.
• It is an open protocol suite. It is not owned by any particular
institute and so can be used by any individual or organization.
• It is a scalable, client-server architecture. This allows networks
to be added without disrupting the current services.
• It assigns an IP address to each computer on the network, thus
making each device to be identifiable over the network. It
assigns each site a domain name. It provides name and address
resolution services.

31
The disadvantages of the TCP/IP model are
• It is not generic in nature. So, it fails to represent any protocol stack other than
the TCP/IP suite. For example, it cannot describe the Bluetooth connection.
• It does not clearly separate the concepts of services, interfaces, and protocols.
So, it is not suitable to describe new technologies in new networks.
• It does not distinguish between the data link and the physical layers, which has
very different functionalities. The data link layer should concern with the
transmission of frames. On the other hand, the physical layer should lay down
the physical characteristics of transmission. A proper model should segregate
the two layers.
• It was originally designed and implemented for wide area networks. It is not
optimized for small networks like LAN (local area network) and PAN (personal
area network).
• Among its suite of protocols, TCP and IP were carefully designed and well
implemented. Some of the other protocols were developed ad hoc and so
proved to be unsuitable in long run. However, due to the popularity of the
model, these protocols are being used even 30–40 years after their
introduction.

32
 UNIT-IV: TRANSPORT LAYER
UDP
1. Segment header
2. Remote Procedure Call (RPC).
3. Real-time Transport Protocols (RTP).
TCP
1. service model.
2. Protocol.
3. Segment header.
4. Connection establishment .
5. Connection release .
6. Sliding window .
7. Timer management.
8. Congestion control. 33
3. The TCP Segment Header

34
1. The Source port and Destination port fields identify the local
end points of the connection. A TCP port plus its host’s IP
address forms a 48-bit unique end point.
The connection identifier is called a 5 tuple because it consists
of five pieces of information: The protocol (TCP), Source IP and
Source Port, and Destination IP and Destination Port.
2. The Sequence number and Acknowledgement number fields
perform their usual functions. Note that the latter specifies the
next in-order byte expected, not the last byte correctly
received. It is a cumulative acknowledgement because it
summarizes the received data with a single number. It does not
go beyond lost data. Both are 32 bits because every byte of
data is numbered in a TCP stream.
3. The TCP header length tells how many 32-bit words are
contained in the TCP header.

35
4. A 4-bit field that is not used.
5. Next comes to eight 1-bit flags:
i. CWR (Congestion Window Reduced)
ii. ECN (Explicit Congestion Notification)
iii. URG (Urgent pointer)
iv. ACK (Acknowledgement number)
v. PSH (PUSHed data)
vi. RST (Reset a connection)
vii. The SYN bit is used to establish connections
viii.The FIN bit is used to release a connection
• ECE (Explicit Congestion Notification-Echo) is set to signal an
ECN-Echo to a TCP sender to tell it to slow down when the
TCP receiver gets a congestion indication from the network.

36
• CWR (Congestion Window Reduced) is set to signal Congestion Window
Reduced from the TCP sender to the TCP receiver so that it knows the
sender has slowed down and can stop sending the ECN-Echo.
• URG is set to 1 if the Urgent pointer is in use. The Urgent pointer is used to
indicate a byte offset from the current sequence number at which urgent
data are to be found. This facility is in lieu of interrupt messages.
• The ACK bit is set to 1 to indicate that the Acknowledgement number is
valid. This is the case for nearly all packets. If ACK is 0, the segment does
not contain an acknowledgement, so the Acknowledgement number field
is ignored.
• The PSH bit indicates PUSHed data. The receiver is hereby kindly
requested to deliver the data to the application upon arrival and not buffer
it until a full buffer has been received (which it might otherwise do for
efficiency).
• The RST bit is used to suddenly reset a connection that has become
confused due to a host crash or some other reason. It is also used to reject
an invalid segment or refuse an attempt to open a connection. In general,
if you get a segment with the RST bit on, you have a problem on your
hands.
37
• The SYN bit is used to establish connections. The connection request
has SYN = 1 and ACK = 0 to indicate that the piggyback
acknowledgement field is not in use. The connection reply does bear an
acknowledgement, however, so it has SYN = 1 and ACK = 1. In essence,
the SYN bit is used to denote both CONNECTION REQUEST and
CONNECTION ACCEPTED, with the ACK bit used to distinguish between
those two possibilities.
• The FIN bit is used to release a connection. It specifies that the sender
has no more data to transmit. However, after closing a connection, the
closing process may continue to receive data indefinitely. Both SYN and
FIN segments have sequence numbers and are thus guaranteed to be
processed in the correct order.
6. Flow control in TCP is handled using a variable-sized sliding window. The
Window size field tells how many bytes may be sent starting at the byte
acknowledged. A Window size field of 0 is legal and Ack. number is 1.

38
7. A Checksum is also provided for extra reliability. It checksums the
header, the data, and a conceptual pseudoheader in exactly the
same way as UDP, except that the pseudoheader has the protocol
number for TCP (6) and the checksum is mandatory.
8. The Options field provides a way to add extra facilities not
covered by the regular header.
The options are of variable length, fill a multiple of 32 bits by
using padding with zeros, and may extend to 40 bytes to
accommodate the longest TCP header that can be specified.
Some options are carried when a connection is established to
negotiate or inform the other side of capabilities. Other options
are carried on packets during the lifetime of the connection. Each
option has a Type-Length-Value encoding. A widely used option
is the one that allows each host to specify the MSS (Maximum
Segment Size). The window scale option allows the sender and
receiver to negotiate a window scale factor at the start of a
connection. 39
• The timestamp option carries a timestamp sent by the sender
and echoed by the receiver.

• Finally, the SACK (Selective ACKnowledgement) option lets a

receiver tell a sender the ranges of sequence numbers that it
has received. It supplements the Acknowledgement number
and is used after a packet has been lost but subsequent
(or duplicate) data has arrived.

40
 UNIT-IV: TRANSPORT LAYER
UDP
1. Segment header
2. Remote Procedure Call (RPC).
3. Real-time Transport Protocols (RTP).
TCP
1. service model.
2. Protocol.
3. Segment header.
4. TCP Connection Establishment .
5. TCP Connection Release .
6. TCP Sliding Window .
7. TCP Timer Management.
8. TCP Congestion Control. 41
 4. TCP Connection Establishment
• Connections are established in TCP by means of the three-way
handshake .
• To establish a connection, one side, say, the server, passively waits
for an incoming connection by executing the LISTEN and ACCEPT
primitives, either specifying a specific source or nobody in
particular.
• The other side, say, the client, executes a CONNECT primitive,
specifying the IP address and port to which it wants to connect,
the maximum TCP segment size it is willing to accept, and
optionally some user data (e.g., a password). The CONNECT
primitive sends a TCP segment with the SYN bit on and ACK bit off
and waits for a response.
• When this segment arrives at the destination, the TCP entity there
checks to see if there is a process that has done a LISTEN on the
port given in the Destination port field. If not, it sends a reply with
the RST bit on to reject the connection 42
 TCP Connection Establishment

6-31

(a) TCP connection establishment in the normal case.

(b) Simultaneous connection establishment on both sides. 43
TCP Connection Management Modeling

The states used in the TCP connection

management finite state machine.

44
 UNIT-IV: TRANSPORT LAYER
UDP
1. Segment header
2. Remote Procedure Call (RPC).
3. Real-time Transport Protocols (RTP).
TCP
1. service model.
2. Protocol.
3. Segment header.
4. Connection establishment .
5. Connection release .
6. Sliding window .
7. Timer management.
8. Congestion control. 45
 5. TCP Connection Release
• Each simplex connection is released independently of its sibling.
• To release a connection, either party can send a TCP segment with
the FIN bit set, which means that it has no more data to transmit.
• When the FIN is acknowledged, that direction is shut down for
new data.
• Data may continue to flow indefinitely in the other direction,
however.
• When both directions have been shut down, the connection is
released.
• To avoid the two-army problem, timers are used.
• If a response to a FIN is not forthcoming within two maximum
packet lifetimes, the sender of the FIN releases the connection.
• The other side will eventually notice that nobody seems to be
listening to it any more and will time out as well.
46
Connection Release: The two-army problem
Unfortunately, this protocol does not always work.
The two-army problem illustrates four scenarios of releasing using
a three-way handshake. While this protocol is not reliable, it is
usually adequate.

47
DR (DISCONNECTION REQUEST)

Four protocol scenarios for releasing a connection.

(a) Normal case of three-way handshake.
(b) Final ACK lost.
(c) Response lost.
(d) Response lost and subsequent DRs lost. 48
 Four protocol scenarios for releasing a connection.
(a) Normal case of three-way handshake.
(b) Final ACK lost.
(c) Response lost.
(d) Response lost and subsequent DRs lost. 49
 UNIT-IV: TRANSPORT LAYER
UDP
1. Segment header
2. Remote Procedure Call (RPC).
3. Real-time Transport Protocols (RTP).
TCP
1. service model.
2. Protocol.
3. Segment header.
4. Connection establishment .
5. Connection release .
6. Sliding window .
7. Timer management.
8. Congestion control. 50
 6. TCP Sliding Window
1. A frame carries an Error-Detecting Code (e.g., a CRC or
checksum) that is used to check if the information was correctly
received.
2. A frame carries a sequence number to identify itself and is
retransmitted by the sender until it receives an
acknowledgement of successful receipt from the receiver. This is
called ARQ (Automatic Repeat reQuest).
3. There is a maximum number of frames that the sender will allow
to be outstanding at any time, pausing if the receiver is not
acknowledging frames quickly enough. If this maximum is one
packet the protocol is called stop-and-wait. Larger windows
enable pipelining and improve performance on long, fast links.
4. The sliding window protocol combines these features and is
also used to support bidirectional data transfer.

51
TCP Sliding Window: TCP Transmission Policy

Window management in TCP. 52

• When the window is 0, the sender may not normally send segments,
with two exceptions. First, urgent data may be sent, for example, to
allow the user to kill the process running on the remote machine.
Second, the sender may send a 1-byte segment to force the receiver to
reannounce the next byte expected and the window size. This packet is
called a window probe.
• The TCP standard explicitly provides this option to prevent deadlock if a
window update ever gets lost.
• One approach that many TCP implementations use to optimize
Bandwidth situation is called delayed acknowledgements.
• The idea is to delay acknowledgements and window updates for up to
500 msec in the hope of acquiring some data on which to catch a free
ride.
• Although delayed acknowledgements reduce the load placed on the
network by the receiver, a sender that sends multiple short packets
(e.g., 41-byte packets containing 1 byte of data) is still operating
inefficiently. A way to reduce this usage is known as Nagle’s algorithm
(Nagle, 1984). 53
• What Nagle suggested is simple: when data come into the sender in
small pieces, just send the first piece and buffer all the rest until the
first piece is acknowledged.
• Then send all the buffered data in one TCP segment and start
buffering again until the next segment is acknowledged.
• That is, only one short packet can be outstanding at any time.
• If many pieces of data are sent by the application in one round-trip
time, Nagle’s algorithm will put the many pieces in one segment,
greatly reducing the bandwidth used.
• The algorithm additionally says that a new segment should be sent if
enough data have trickled in to fill a maximum segment.
• A more subtle problem is that Nagle’s algorithm can sometimes
interact with delayed acknowledgements to cause a temporary
deadlock: the receiver waits for data on which to piggyback an
acknowledgement, and the sender waits on the acknowledgement to
send more data. This interaction can delay the downloads of Web
pages. Because of these problems, Nagle’s algorithm can be disabled54
• Another problem that can degrade TCP performance is the
Silly window syndrome (Clark, 1982). This problem occurs when
data are passed to the sending TCP entity in large blocks, but an
interactive application on the receiving side reads data only 1 byte
at a time.
• Clark's solution is to prevent the receiver from sending a window
update for 1 byte.
• Instead it is forced to wait until it has a decent amount of space
available and advertise that instead.
• Specifically, the receiver should not send a window update until it
can handle the maximum segment size it advertised when the
connection was established or until its buffer is half empty,
whichever is smaller

55
• Nagle’s algorithm and Clark’s solution to the silly window
syndrome are complementary.
• Nagle was trying to solve the problem caused by the sending
application delivering data to TCP a byte at a time.
• Clark was trying to solve the problem of the receiving application
sucking the data up from TCP a byte at a time. Both solutions are
valid and can work together.
• The goal is for the sender not to send small segments and the
receiver not to ask for them.
• Acknowledgements can be sent only when all the data up to the
byte acknowledged have been received. This is called a
cumulative acknowledgement.

56
Silly window syndrome. 57
 UNIT-IV: TRANSPORT LAYER
UDP
1. Segment header
2. Remote Procedure Call (RPC).
3. Real-time Transport Protocols (RTP).
TCP
1. service model.
2. Protocol.
3. Segment header.
4. Connection establishment .
5. Connection release .
6. Sliding window .
7. Timer management.
8. Congestion control. 58
 7. TCP Timer Management
• TCP uses multiple timers to do its work.
– Retransmission timer
– Persistence timer
– Keepalive timer
• The most important of these is the RTO (Retransmission
TimeOut). When a segment is sent, a retransmission timer is
started.
• If the segment is acknowledged before the timer expires, the
timer is stopped.
• If, on the other hand, the timer goes off before the
acknowledgement comes in, the segment is retransmitted.
• The question that arises is: How long should the timeout
interval be?
59
• As Shown in Figure (a) Probability density of acknowledgement
arrival times in the data link layer.
• Since acknowledgements are rarely delayed in the data link layer
(due to lack of congestion), the absence of an acknowledgement
at the expected time generally means either the frame or the
acknowledgement has been lost.
• If the timeout is set too short, say, T1 in Figure (b), unnecessary
retransmissions will occur, clogging the Internet with useless
packets.
• If it is set too long (e.g., T2), performance will suffer due to the
long retransmission delay whenever a packet is lost. Furthermore,
the mean and variance of the acknowledgement arrival
distribution can change rapidly within a few seconds as congestion
builds up or is resolved.

60
(a) Probability density of acknowledgement arrival times in the data
link layer.
(b) Probability density of acknowledgement arrival times for TCP.
61
• The solution is to use a dynamic algorithm that constantly adapts
the timeout interval, based on continuous measurements of
network performance.
• The algorithm generally used by TCP is due to Jacobson (1988) and
works as follows. For each connection, TCP maintains a variable,
SRTT (Smoothed Round-Trip Time), that is the best current
estimate of the round-trip time to the destination in question.
• When a segment is sent, a timer is started, both to see how long
the acknowledgement the acknowledgement gets back before the
timer expires, TCP measures how long the acknowledgement took,
say, R. It then updates SRTT according to the formula
SRTT = α SRTT + (1 − α) R
• where α is a smoothing factor that determines how quickly the old
values are forgotten.
• Typically, α = 7/8. This kind of formula is an EWMA (Exponentially
Weighted Moving Average) or low-pass filter that discards noise in
the samples. 62
• RTTVAR (Round-Trip Time VARiation) that is updated using the
formula
RTTVAR = β RTTVAR + (1 − β) |SRTT − R |
• This is an EWMA as before, and typically β = 3/4.
• The retransmission timeout, RTO, is set to be
RTO = SRTT + 4 × RTTVAR

63
 Persistence timer
• Phil Karn made a simple proposal: do not update estimates on any
segments that have been retransmitted. Additionally, the timeout
is doubled on each successive retransmission until the segments
get through the first time. This fix is called Karn’s algorithm (Karn
and Partridge, 1987). Most TCP implementations use it.
• A second timer is the persistence timer. It is designed to prevent
the following deadlock.
• The receiver sends an acknowledgement with a window size of 0,
telling the sender to wait. Later, the receiver updates the window,
but the packet with the update is lost. Now both the sender and
the receiver are waiting for each other to do something.
• When the persistence timer goes off, the sender transmits a probe
to the receiver. The response to the probe gives the window size.
If it is still zero, the persistence timer is set again and the cycle
repeats. If it is nonzero, data can now be sent. 64
 Keepalive timer
• A third timer that some implementations use is the keepalive
timer.
• When a connection has been idle for a long time, the keepalive
timer may go off to cause one side to check whether the other
side is still there.
• If it fails to respond, the connection is terminated.
• This feature is controversial because it adds overhead and may
terminate an otherwise healthy connection due to a transient
network partition.
• The last timer used on each TCP connection is the one used in the
TIME WAIT state while closing.
• It runs for twice the maximum packet lifetime to make sure that
when a connection is closed, all packets created by it have died
off.
65
 UNIT-IV: TRANSPORT LAYER
UDP
1. Segment header
2. Remote Procedure Call (RPC).
3. Real-time Transport Protocols (RTP).
TCP
1. service model.
2. Protocol.
3. Segment header.
4. Connection establishment .
5. Connection release .
6. Sliding window .
7. Timer management.
8. Congestion control. 66
 8. TCP Congestion Control
• When the load offered to any network is more than it can handle,
congestion builds up.
• The Internet is no exception. The network layer detects
congestion when queues grow large at routers and tries to
manage it, if only by dropping packets.
• It is up to the transport layer to receive congestion feedback from
the network layer and slow down the rate of traffic that it is
sending into the network.
• In the Internet, TCP plays the main role in controlling congestion,
as well as the main role in reliable transport. That is why it is such
a special protocol.

67
• A transport protocol using an AIMD (Additive Increase
Multiplicative Decrease) control law in response to binary
congestion signals from the network would converge to a fair and
efficient bandwidth allocation.
• TCP congestion control is based on implementing this approach
using a window and with packet loss as the binary signal.
• To do so, TCP maintains a congestion window whose size is the
number of bytes the sender may have in the network at any time.
• The corresponding rate is the window size divided by the round-
trip time of the connection.
• TCP adjusts the size of the window according to the AIMD rule.

68
• Before discussing how TCP reacts to congestion, let us first
describe what it does to try to prevent congestion from
occurring in the first place.
• When a connection is established, a suitable window size has
to be chosen.
• The receiver can specify a window based on its buffer size.
• If the sender sticks to this window size, problems will not
occur due to buffer overflow at the receiving end, but they
may still occur due to internal congestion within the network.

69
• The below figure shows what happens when a sender on a fast network (the 1-
Gbps link) sends a small burst of four packets to a receiver on a slow network (the
1- Mbps link) that is the bottleneck or slowest part of the path.
• Initially the four packets travel over the link as quickly as they can be sent by the
sender.
• At the router, they are queued while being sent because it takes longer to send a
packet over the slow link than to receive the next packet over the fast link.
• But the queue is not large because only a small number of packets were sent at
once. Note the increased length of the packets on the slow link.
• The same packet, of 1 KB say, is now longer because it takes more time to send it
on a slow link than on a fast one.

A burst of packets from a sender and the returning ack clock. 70

• the acknowledgements return to the sender at about the rate that
packets can be sent over the slowest link in the path. This is
precisely the rate that the sender wants to use.
• If it injects new packets into the network at this rate, they will be
sent as fast as the slow link permits, but they will not queue up and
congest any router along the path.
• This timing is known as an ack clock. It is an essential part of TCP. By
using an ack clock, TCP smoothes out traffic and avoids unnecessary
queues at routers.
• A second consideration is that the AIMD rule will take a very long
time to reach a good operating point on fast networks if the
congestion window is started from a small size. It would cause
congestion if used all at once.
• The solution Jacobson chose to handle both of these considerations
is a mix of linear and multiplicative increase.

71
• When a connection is established, the sender initializes the congestion
window to a small initial value of at most four segments; and the use of
four segments is an increase from an earlier initial value of one segment
based on experience.
• The sender then sends the initial window. The packets will take a round-
trip time to be acknowledged. For each segment that is acknowledged
before the retransmission timer goes off.
• The congestion window is doubling every Round Trip Time.
• The slow start algorithm, but it is not slow at all—it is exponential
growth—except in comparison to the previous algorithm that let an
entire flow control window be sent all at once.
• In the first round-trip time, the sender injects one packet into the
network (and the receiver receives one packet). Two packets are sent in
the next round-trip time, then four packets in the third round-trip time.
• Slow-start works well over a range of link speeds and round-trip times,
and uses an ack clock to match the rate of sender transmissions to the
network path.
72
Fig. 6-44 Slow start algorithm from an initial congestion window of one
segment.
73
• Slow Start causes exponential growth, eventually (and sooner
rather than later) it will send too many packets into the network
too quickly. When this happens, queues will build up in the
network.
• When the queues are full, one or more packets will be lost. After
this happens, the TCP sender will time out when an
acknowledgement fails to arrive in time.
• There is evidence of slow start growing too fast in Fig. 6-44. After
three RTTs, four packets are in the network. These four packets
take an entire RTT to arrive at the receiver.
• Additional packets placed into the network by the sender will
build up in router queues, since they cannot be delivered to the
receiver quickly enough.
• Congestion and packet loss will occur soon.
• To keep slow start under control, the sender keeps a threshold for
the connection called the slow start threshold.
74
• To keep slow start under control, the sender keeps a threshold for the connection
called the slow start threshold. Initially this value is set arbitrarily high, to the size
of the flow control window, so that it will not limit the connection.
• TCP keeps increasing the congestion window in slow start until a timeout occurs
or the congestion window exceeds the threshold.
• Whenever a packet loss is detected, for example, by a timeout, the slow start
threshold is set to be half of the congestion window and the entire process is
restarted.
• Whenever the slow start threshold is crossed, TCP switches from slow start to
additive increase.
• Call the congestion window cwnd and the maximum segment size MSS.
• A common approximation is to increase cwnd by (MSS × MSS)/cwnd for each of
the cwnd /MSS packets that may be acknowledged. This increase does not need
to be fast.
• The whole idea is for a TCP connection to spend a lot of time with its congestion
window close to the optimum value—not so small that throughput will be low,
and not so large that congestion will occur.
• Additive increase is shown in below figure, for the same situation as slow start. At
the end of every RTT, the sender’s congestion window has grown enough that it
can inject an additional packet into the network. 75
Additive increase from an initial congestion window of one segment. 76
• Drawback: Timeouts are relatively long because they must be
conservative. After a packet is lost, the receiver cannot
acknowledge past it, so the acknowledgement number will stay
fixed, and the sender will not be able to send any new packets into
the network because its congestion window remains full.
• This condition can continue for a relatively long period until the
timer fires and the lost packet is retransmitted. At that stage, TCP
slow starts again.
• There is a quick way for the sender to recognize that one of its
packets has been lost. As packets beyond the lost packet arrive at
the receiver, they trigger acknowledgements that return to the
sender.
• These acknowledgements bear the same acknowledgement
number. They are called duplicate acknowledgements.

77
• TCP somewhat arbitrarily assumes that three duplicate
acknowledgements imply that a packet has been lost. The identity
of the lost packet can be inferred from the acknowledgement
number as well.
• It is the very next packet in sequence. This packet can then be
retransmitted right away, before the retransmission timeout fires.
• This heuristic is called fast retransmission.
• Fast recovery is a temporary mode that aims to maintain the ack
clock running with a congestion window that is the new threshold,
or half the value of the congestion window at the time of the fast
retransmission.

78
 sawtooth pattern of additive increase
• The upshot of this heuristic is that TCP avoids slow start,
except when the connection is first started and when a
timeout occurs.
• The latter can still happen when more than one packet is lost
and fast retransmission does not recover adequately.
• Instead of repeated slow starts, the congestion window of a
running connection follows a sawtooth pattern of additive
increase (by one segment every RTT) and multiplicative
decrease (by half in one RTT).
• This is exactly the AIMD rule that we sought to implement.

79
• The sawtooth behavior is shown in below figure.

Fast recovery and the sawtooth pattern of TCP Reno.

80
• The cumulative acknowledgement number does not provide the
information of which packets have arrived and which packets
have been lost.
• A simple fix is the use of SACK (Selective ACKnowledgements),
which lists up to three ranges of bytes that have been received.
With this information, the sender can more directly decide what
packets to retransmit and track the packets in flight to implement
the congestion window.
• When the sender and receiver set up a connection, they each
send the SACK permitted TCP option to signal that they
understand selective acknowledgements.

81
Selective acknowledgements

82
 The Future of TCP
• The first issue with TCP is : TCP does not provide the transport
semantics that all applications want. For example, some
applications want to send messages or records whose boundaries
need to be preserved. Other applications work with a group of
related conversations, such as a Web browser that transfers several
objects from the same server.
• Still other applications want better control over the network paths
that they use. TCP with its standard sockets interface does not
meet these needs well.
• Essentially, the application has the burden of dealing with any
problem not solved by TCP.
• This has led to proposals for new protocols that would provide a
slightly different interface. Two examples are SCTP (Stream Control
Transmission Protocol), defined in RFC 4960, and SST (Structured
Stream Transport) (Ford, 2007).
83
• The second issue is congestion control.
• The signal might be round-trip time, which grows when the
network becomes congested, as is used by FAST TCP (Wei et al.,
2006). Other approaches are possible too, and time will tell which
is the best.

84