Congestion Avoidance Mechanism by Multiple Routers

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International Journal of Engineering Trends and Technology (IJETT) – Volume 4 Issue 9- Sep 2013
Congestion Avoidance Mechanism by Multiple
Routers
Sailaja.Sanaka#1, Narasimha Rao PDL#2, Neelima Priyanka Nutulapati#3
1
Assistant Professor, CSE Department, SRK Institute of Technology
Assistant Professor, CSE Department, SRK Institute of Technology
3
Assistant Professor, CSE Department, SRK Institute of Technology
Vijayawada, Andhra Pradesh, India
2
Abstract— Goal of any congestion control technique is to avoid
congestion in any network. Designing and implementing such
techniques would be more tedious and may require additional
resources. Typically when too many packets are transmitted
through a network, congestion occurs. To avoid data loss in such
situations we need proactive mechanism to prevent blockages.
Here we propose a new congestion avoidance mechanism by
means of hierarchical routers. Primary router uses packet
switching from one router to another which yields reduced
packet drops. Generally if the sender does not receive an
acknowledgment before the timeout, it usually retransmits the
same packet. The proposed hierarchical routing technique helps
to decrease such type of retransmission. We report on an initial
exploration of this idea and gives future directions for further
research.
Keywords— Congestion avoidance, Hierarchical routers, Packet
switching.
I. INTRODUCTION
Now-a-days data travels from one network to
another where data transfer done through internet.
Essentially the router is a switching element which
processes messages over one link onto another link
for transmission to an end system or another router.
When too many packets are transmitted through a
network, congestion occurs. Network congestion
avoidance techniques are important in-order to send
data successfully.
II. PRESENT SYSTEM
A. Flow control
Before the source starts sending packets, it will
send a request to the destination for permission to
start transmission. In response to the request, the
destination sends a message containing an
identification of the number of packets the source
may dispatch towards destination without further
authorization. This number is commonly known as
window size. The source then proceeds to transmit
the authorized number of packets towards the
destination and waits for the destination reply. After
the destination successfully receives a packet it
sends a message back to the source containing an
acknowledgement indicating the successful receipt
of the packet and in some cases, authorizing the
source to send another packet. In this way the
number of packets on the network travelling from
the source towards the destination will never be
more than the authorized window size.
B. Usage of buffer
An important design objective in networks is
controlling the flow of packets so that they will not
be transmitted at a faster rate than they can be
processed by the routers. Even in the network
consisting of two end systems interconnected by a
router, the source may flood the destination if it
transmits packets faster than they can be processed
by the destination. In the networks consisting of
many end systems, more number of routers and
ISSN: 2231-5381
alternative communication paths between the end
systems, the likelihood of problems from excess
communication traffic is significantly more.
The main mechanisms to deal with excess traffic
in the network are flow control and usage of buffers.
In buffered systems, buffer memory is provided
for the routers and the end systems to handle
overloads. Packets which exceeds processing rate
of the device are temporarily stored in the buffer
memory until the device can process it. Buffer
system show a satisfactory solution to excess traffic
problems only if the overload is transitory. If the
buffer is overloaded the additional packets are
rejected or destroyed. If the packet is dropped the
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International Journal of Engineering Trends and Technology (IJETT) – Volume 4 Issue 9- Sep 2013
source resends the same packet to the router which
is already congested. To avoid such packet
dropping and resending, this paper is a solution.
C. Disadvantages
All these protocols usually use sliding
window protocol and identify packets to be
retransmitted. These protocols reside in the
Data Link or Transport Layers of the OSI
model.
1) If the router is overloaded, the buffers may C. Dijkstra’s algorithm
become full after which the additional packets
are rejected or destroyed.
For a given graph this algorithm finds the
2) If a packet is dropped, the source resends the
shortest path from a source node to the
same packet to the router which is already
destination node with lowest cost between
congested.
that vertex and every other vertex.
III. RELATED WORK
All networks today are facing this congestion
problem which needs a serious solution to improve
the efficiency.
A. Multicasting
A number of emerging network applications
require the delivery of packets from one or more
senders to a group of receivers. These applications
include bulk data transfer, transfer of audio, video
and text of a live program, shared data applications
such as teleconferencing, a white board etc., data
feeds such as multiplayer games. All these
applications need multicasting which sends data
from one sender to multiple receivers with a single
transmit operation. Multicast is emulated using
multiple point-to-point uni-cast connections.
B. Automatic repeat request(ARQ)s
Automatic repeat request is also known as
automatic repeat query. It is an error control method
for data transmission that uses acknowledgements.
Acknowledgements are messages sent by the
receiver indicating that it has correctly received a
data packet for reliable data transmission. If the
sender does not receive an acknowledgement before
the timeout, it usually retransmits the frame until
the sender receives an acknowledgement.
The types of ARQ protocols include
1. Stop-and-wait ARQ
2. Go-Back-N ARQ
3. Selective Repeat ARQ
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Let the node at which we are starting be
called the initial node. Let the distance node
Y be the distance from the initial node to Y.
Dijkstra’s algorithm will assign some initial
distance values and will try to improve them
step by step.
a) Every node should be assigned a tentative
distance value. Set it to zero for our initial
node and to infinity for all other nodes.
b) Mark all nodes unvisited. Set the initial
node as current. Create a set of the
unvisited nodes. The set should contain all
the nodes except the initial node.
c) Calculate unvisited node’s tentative
distances from the current node.
d) When we have finished considering all of
the neighbours of the current node, mark
the current node as visited and remove it
from the unvisited set. A visited node will
never be checked again.
e) If the destination node has been marked as
visited (when we are planning a route
between two specific nodes) or if the
smallest tentative distance among the
nodes in the unvisited set is infinity then
stop. The algorithm has finished.
f) Select the unvisited node that is marked
with the smallest tentative distance and
set it as the new current node then go back
to step3.
IV. PROPOSED SYSTEM
A. Retransmission
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International Journal of Engineering Trends and Technology (IJETT) – Volume 4 Issue 9- Sep 2013
Generally the time-to-live (TTL) value can be
thought of as an upper bound on the time that a
datagram can exist in the internet system. The TTL
field is set by the sender of the datagram, and
reduced by every router on the route to its
destination. If the TTL field reaches zero before the
datagram arrives at its destination, then the
datagram is discarded and an ICMP error datagram
is sent back to the sender. The purpose of the TTL
is to avoid a situation in which an undeliverable
datagram keeps circulating on an internet system
and such a system eventually swamped by such
“immortals”.
Under the internet protocol TTL is an 8-bit field.
In the IPV4 header, TTL is the 9 th octet of 20. In the
IPV6 header, it is the 8th octet of 40. The maximum
TTL value is 255, the maximum value of a single
octet. A recommended initial value is 64.
Retransmission is a very simple concept.
Whenever one party sends something to the other
party, it retains a copy of the data it sent until the
recipient has acknowledged that it received it. In a
variety of circumstances the sender automatically
retransmits the data using the retained copy.
Reasons for resending include:
If no such acknowledgments forthcoming within
a reasonable time, the time-out
The sender discovers often through some out of
band means, that the transmission was unsuccessful
If the receiver knows that expected data has not
arrived, and so notifies the sender.
If the receiver knows that expected data has
arrived, but in a damaged condition, and indicates
that to the sender.
For a number of reasons, packets may not get
delivered to their destination in a reasonable length
of time. For example, a combination of incorrect
routing tables could cause a packet to loop
endlessly. A solution is to discard the packet after a
certain time and send a message to the originator,
who can decide whether to resend the packet. The
initial TTL value is set, usually by a system defaults,
in an 8-binary digit field of the packet header. The
original idea of TTL was that it would specify a
certain time span in seconds that, when exhausted,
would cause the packet to be discarded. Since each
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router is required to subtract at least one count from
the TTL field, the count is usually used to mean the
number of router hops the packet is allowed before
it must be discarded. Each router that receives a
packet subtracts one from the count in the TTL filed.
When the count reaches zero, the router detecting it
discards the packet and sends an internet control
message protocol message back to the originating
host.
The ping and trace route utilities both make use
of TTL value to reach a given host computer or to
trace a route to that host. Trace route intentionally
sends a packet with a low TTL value so that it will
be discarded by each successive router in the
destination path. The time between sending the
packet and receiving back the ICMP message that it
was discarded is used to calculate each successive
hop travel time.
Using the multicast IP protocol, the TTL value
indicates the scope or range in which a packet may
be forwarded. By convention:
1) 0 is restricted to the same host
2) 1 is restricted to the same subnet
3) 32 is restricted to the same site
4) 64 is restricted to the same region
5) 128 is restricted to the same continent
6) 255 is unrestricted
B. Architecture
The invention provides a new and improved
mechanism and associated method for avoiding
congestion on a network. In this paper we propose
the hierarchical routers to decrease packet dropping
by the router and to reduce resending of packets by
source with the mechanism of redirecting the
packets to nearby routers or nodes whose buffer is
free. Here source sends data packets to the main
router. If main router’s buffer is below the pre-set
threshold value, main router itself sends it to the
destination directly. If the buffer size is equal or
greater than the threshold value then main router
will send a request to the nearest routers or nodes to
know their window size. In response to the request,
the sub routers those who are willing, will send a
message containing an identification of the number
of packets the source may dispatch towards the
destination without further authorization. This
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number is commonly referred to as the window size.
Then the main router selects a sub router or node by
applying Dijkstra’s algorithm and redirects the data
packets to that sub router. This avoids packet
dropping. Router also informs the source to
increase the life time of the next packets to avoid
resending of packets. This decreases the packet
resending. In this way congestion may be avoided
at the router and retransmission at source is also
decreased.
The following is the architecture of congestion
avoidance:
Fig. 2 Flow for the congestion avoidance mechanism
V.
Fig.1 Architecture for congestion avoidance mechanism
CONCLUSION
Designing an efficient router for the future
internet is a challenging task. Our design is a two
way congestion avoidance technique. When a
packet is dropped at the router because of
congestion, the source has to resend it which also
increases the congestion. Our design shows two
ways two to decrease this congestion. One way is
by placing a threshold value to the buffer we can
avoid congestion and the other way is
retransmission of the packet by the sender is
minimized by increasing the life time of the packet.
Retransmission and packet loss would be low in our
design. Compared with other proposals this can be
easier to implement.
ACKNOWLEDGMENT
The following is the flow of this mechanism:
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This research paper would not have been possible
without the support of many people. The author
wishes to express gratitude to A.Radhika, Senior
Assistant Professor in SRK Institute of Technology
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who was abundantly helpful and offered invaluable
assistance, support and guidance. The author would
also like to convey thanks to the SRK Institute of
Technology for providing unperturbed internet
facilities.
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