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International Research Journal of Engineering and Technology (IRJET) e-ISSN: 2395-0056
Volume: 06 Issue: 03 | Mar 2019 www.irjet.net p-ISSN: 2395-0072
© 2019, IRJET | Impact Factor value: 7.211 | ISO 9001:2008 Certified Journal | Page 7563
QOS BASED BANDWIDTH SATISFACTION FOR MULTICAST NETWORK
CODING IN MANET
C.Shalini1 , K.Senthil Prakash2
1. Second Year-M.E. (Applied Electronics), Velalar College of Engineering and Technology, Erode.
2. Assistant Professor(Sl.Gr), Department of ECE, Velalar College of Engineering and Technology, Erode.
-----------------------------------------------------------------------------***---------------------------------------------------------------------------
ABSTRACT-Wireless links are lossy due to network traffic
and fading channel. To avoid the packet loss in this method
duplicate packet is forwarded in the link to get the
acknowledgement from a receiver. Multipath routing trees
are constructed for the transmission which providing the
bandwidth guarantee. With the randomized network coding,
the probability of overhearing the same packet transmission
by different forwarder is very low. The proposed multicast
routing protocol is designed based on contention-based
media access control (MAC) protocols. Scheduling algorithm
is not separately needed for distributing packets among the
constructed trees. Estimation of the bandwidth before
transmission is obtained using a variable bit rate. To
improve the accuracy of system and earlier estimation of the
bandwidth, multicast routing protocols are used. This
method reduced the delay and retransmission rates and
improve the bandwidth usage in the MANET.
Key Words: MANET, QOS, HRP, HMRP.
1. INTRODUCTION
A mobile ad hoc network (MANET) enables
wireless communications between participating mobile
nodes without the assistance of any base station. Two
nodes that are out of one another’s transmission range to
need the support of intermediate nodes, which a relay
messages to set up a communication between each other.
The most fundamental role in MANET is the broadcast
operation because of the broadcasting nature of radio
transmission: When a sender transmits a packet, all nodes
within the sender’s transmission range will be affected by
this transmission. The advantage is if one node transmits a
packet, all its neighbors can receive this message. In the
negative side, one transmission may interfere with other
transmissions, creating the exposed terminal problem
where an outgoing transmission collides with an incoming
transmission and the hidden terminal problem where two
incoming transmissions collide with each other.
Some existing QoS-guaranteed multicast protocols for
MANETs assume ideal links. However, links are lossy in
real wireless networks. The packet loss ratio, which
represents the probability that a packet is lost or
erroneous, depends on the channel quality. When the
packet loss ratio of a lossy link increases, more bandwidth
consumption for retransmission is required .Since real
wireless links are lossy, it is practically important to
consider lossy MANETs when we are designing a
bandwidth-satisfied multicast protocol. It is a great
challenge to design a multicast routing protocol for lossy
MANETs that can provide bandwidth guarantees, avoid
redundant packets, and reduce the bandwidth
consumption at the same time. Providing bandwidth
guarantees to traffic flows may incur lots of redundant
packet due to lost packets. No bandwidth-satisfied
unicast/multicast routing protocol for lossy MANETs was
proposed MANETs suffer from a high transmission error
rate because of the high transmission contention and
congestion. Furthermore, it is a major challenge to provide
high reliability for broadcasting operations under such
dynamic MANETs.
2. EXISTING SYSTEM
In RACC mechanism, the receiver not only
performs the function of flow control, but also participates
in the congestion control. It first measures the bandwidth,
and then computes an appropriate congestion window size
based on the measured bandwidth and the RTT. To
perform these functions, the receiver has to maintain two
timers: one timer for recording the packet inter-arrival
interval and the other for measuring the RTT. The sender
makes use of this information about its receiver to adjust
the congestion window (Xu 2006).
2.1 FUNCTION OF RECEIVER
The bandwidth is measured from the receiver
according to the packet inter-arrival interval. This method
International Research Journal of Engineering and Technology (IRJET) e-ISSN: 2395-0056
Volume: 06 Issue: 03 | Mar 2019 www.irjet.net p-ISSN: 2395-0072
© 2019, IRJET | Impact Factor value: 7.211 | ISO 9001:2008 Certified Journal | Page 7564
can remedy the oscillation in the estimation of bandwidth
of TCP Westwood (Casetti 2002).
Let Bw be the measured bandwidth, L be the data
packet size, and tint be the packet inter-arrival interval.
Then, one can estimate the available bandwidth by
Bw =L/tint for each packet arrival. Moving average method
is used within each congestion window. Let Bw be the i-th
measured bandwidth. Then, the bandwidth can be
continuously updated by below the equation (3.1).
Bw=αBw+(1-α)Bwi (3.1)
Where is an exponential filter co-efficient,
=0.9 is a good value because the former averaged values
should have a higher weight (0.9 in this mechanism) to
lower the measure variations. To reduce the consequence
of cross traffic, a low pass filter can also be adopted.
The receiver measures the RTT based on the data
packet arrival time. A variable rcv.rtt is used to record the
RTT. The algorithm can be described as follows.
i) When an ACK is sent, if rcv.rtt is zero, let rcv.rtt equal
1 and record the corresponding sequence (rtseq) of data
packets (equals to the sum of the ACK sequence and the
current congestion window). Otherwise, just send the ACK.
ii) If a data packet of a larger sequence number than
“rtseq” is arrived in order and the new measured packet
inter-arrival interval is larger than two times of pre-
estimated packet inter-arrival intervals, set the new
measured RTT to the value of rcv.rtt, and let rcv.rtt be zero.
iii) On each clock cycle, if rcv.rtt is not zero, rcv.rtt is
added by 1. Then, the retransmission timer is setting based
on the measured RTT.
Next, the receiver uses RTT to convert the
bandwidth to the receiver congestion window value rwnd
using the below equation (3.2)
rwnd=Bw*RTT (3.2)
Then, it analyses the rwnd with the available
receiver buffer, and deposits the lesser value
of the advertised window field of an ACK going
back to the sender. It is shown in the below equation (3.3)
adv_wnd=min(r_abuf,rwnd) (3.3)
In other words, the receiver advertised window
not only has the original flow control function, yet also
takes on the congestion control function. When the
receiver detects a timeout, it will send an ACK to and
inform the sender that the network is congested. In this
ACK, the receiver advertised window is set to one packet,
meaning that the sender must retransmit the dropped
packet. As the receiver must have detected this timeout
earlier than the sender, it will help the sender to reduce
the waiting time and confirm the packet loss.
2.1.1 MANAGING THE RTO TIMER
An implementation should manage the
retransmission timer(s) in such a way that a segment is
never retransmitted too early, i.e. less than one RTO after
the previous transmission of that segment.
The following algorithm is recommended for
managing the retransmission timer:
1. Every time a packet containing data is sent, if the
timer is not running, start it running so that it will
expire after RTO seconds (for the current value of
the RTO).
2. When much outstanding data has been
acknowledged, turn off the retransmission timer.
3. When an ACK is received that recognizes new
data, restart the retransmission timer so that it
will expire after RTO seconds.
When the retransmission timer expires, do the following:
4. Retransmits the earliest segment that has not
been acknowledged by the TCP receiver.
5. The host must set RTO <- RTO * 2 ("back off the
timer"). The maximum value may be used to
provide an upper bound to this doubling
operation.
6. Start the retransmission timer, such that it expires
after RTO seconds (for the value of RTO after the
doubling operation.
The TCP implementation may clear SRTT and
RTTVAR after backing off the timer multiple times as it is
likely that the current SRTT and RTTVAR are bogus in this
situation. Once SRTT and RTTVAR are cleared they should
be initialized with the next RTT sample taken.
2.2 REACTION OF SENDER
As the receiver can timely help the sender to
increase the congestion window according to the
instantaneous available bandwidth, the sender only needs
to maintain the AIMD mechanism in the congestion
avoidance stage, and the slow start stage can be
eliminated. Upon a timeout event (whether it is detected
by the sender’s timer or informed by the receiver’s ACK),
the sender will decrease the congestion window to one in
consideration that the network is in congestion (Srinivas
2010) and it will have some time to recover. If congestion
is mitigated after one RTT, the sender will recover to
adjust the congestion window in the next window by using
the receiver advertised window. During fast
retransmission, the sender sets the congestion window
size of the lesser value of the receiver advertises window
size and the current size. Since the packet loss may also
indicate congestion, we should reduce the congestion
 
International Research Journal of Engineering and Technology (IRJET) e-ISSN: 2395-0056
Volume: 06 Issue: 03 | Mar 2019 www.irjet.net p-ISSN: 2395-0072
© 2019, IRJET | Impact Factor value: 7.211 | ISO 9001:2008 Certified Journal | Page 7565
window. On the other hand, if the congestion window is
less than the receiver advertised window, the network
may only be in a mild congestion (Afanasyev 2010).
Therefore, it is unnecessary for the sender to
reduce the congestion window. In a normal state, when the
sender receives an ACK, it will compare the current
congestion window with the receiver advertised
congestion window. If the receiver advertised window is
larger than the sender’s congestion window, and the
difference is larger than a predefined threshold, the sender
will set the congestion window size to the receiver
advertised window. Or else, the sender will ignore the
receiver advertised congestion window by just performing
the additive increase mechanism. The threshold value of
TCP Vegas (Brakmo 2001) is used.
3.METHODOLOGY
A multicast routing protocol for lossy MANETs is
proposed. By means of constructing a multiple multicast
trees (trees for short) and transmitting coded packets of
randomized network coding the proposed multicast
routing protocol can provide bandwidth guarantees, avoid
redundant packets, and reduce the total bandwidth
consumption to a certain degree. With the randomized
network coding, the probability that two forwarders
overhearing the same packets transmit the same coded
packets is very low.
The proposed multicast routing protocol is
designed based on the contention-based media access
control (MAC) protocols. Scheduling algorithm is not
separately needed for distributing packets of the
constructed trees. In order to provide bandwidth
guarantees to multicast destinations, the bandwidth that
each source-to-destination route can provide must be
aware. The forwarders can provide the minimum
bandwidth for the route. To determine the bandwidth that
a forwarder can provide, its residual bandwidth and
bandwidth consumption are estimated by the proposed
multicast routing protocol. Moreover, its one-hop and two-
hop neighbors are also examined in order to prevent the
hidden route problem (HRP) and the hidden multicast
route problem (HMRP)
3.1 NEW BANDWIDTH-SATISFIED MULTICAST
ROUTING PROTOCOL
In this section, we intend to propose a multicast
routing protocol for lossy MANETs that can reduce the
total bandwidth consumption to a certain degree while
providing bandwidth guarantees to a requested flow and
ongoing flows. Multiple multicast trees are constructed so
that the residual bandwidth can be fully utilized and the
bandwidth consumption can be reduced. In addition, the
randomized network coding is applied so that redundant
packets can be avoided and each destination can receive
innovative coded packets of distinct routes in the proposed
multicast routing protocol.
A tree construction algorithm is proposed to
construct a multicast tree at a time. It selects hosts of
better channel conditions as forwarders to reduce the total
bandwidth consumption. Each constructed tree connects
some destinations, and each can provide a predefined
percentage of the bandwidth requirement of each
connected destination. The tree construction will continue
until all destinations are bandwidth satisfied. There is a
basic procedure contained in the proposed algorithm.
3.2 TREE CONSTRUCTION ALGORITHM
Given a requested flow Γ, the tree construction
algorithm can construct one or more trees. Sequentially, to
provide bandwidth guarantees to the destinations of Γ and
to reduce the total bandwidth consumption. There are four
input parameters, i.e., hs, D, b ¨ _req, and b_per, for the
algorithm, where hs is the source, D¨ is the set of all
destinations of Γ, b_req is the bandwidth requirement of
each destination, and b_per is a predefined percentage for
the bandwidth requirement (i.e., b_req). There are three
sets D, D , D of destinations, one set F of forwarders, and a
series of 2-D arrays B1, B2,...,Bϕ, B˜ used in the algorithm,
where ϕ is the number of trees constructed. Given a host
hi and a destination hd, Bt [i, d] records the bandwidth of
hi that is consumed for hd in the tth constructed tree,
where 1 ≤ t ≤ ϕ, whereas B˜[i, d] accumulates the
bandwidth of hi that is consumed for hd in all constructed
trees.
3.3 ESTIMATION OF AVAILABLE BANDWIDTH
The available bandwidth estimation is needed in
wireless network because there is no control in the flow of
data and gets affected due to the interference such as
network traffic, link failures and the channel may either
over-utilized or underutilized. Therefore, a deterministic
approach to provisioning service generally overestimates
the actual resource needs, resulting in a low utilization of
network resources. The majority of QOS solutions are
loaded with capability for measuring the jitter, delay,
available bandwidth. The term available bandwidth can be
defined as the maximum throughput that can be
transmitted on one link or between two neighbor nodes
without influence on any existing flow in the network. The
bandwidth reservation solutions designed to 802.11 based
International Research Journal of Engineering and Technology (IRJET) e-ISSN: 2395-0056
Volume: 06 Issue: 03 | Mar 2019 www.irjet.net p-ISSN: 2395-0072
© 2019, IRJET | Impact Factor value: 7.211 | ISO 9001:2008 Certified Journal | Page 7566
ad hoc networks use stochastic estimations of available
bandwidth. Stochastic bandwidth estimations of random
service are based on inferring an unknown bounding
function of measurements of variable bit rates (VBR)
probing traffic. It is one of the processes to monitor the
channel usage based on the CSMA scheme for IEEE 802.11.
However, all the existing approaches do not consider the
random factors in bandwidth estimations. The proposed
system is to improve the accuracy of available bandwidth
estimations by considering the random factors such as
incoming rates; outgoing rate and packets type
consideration (e.g., control packets and data packet).
4.SIMULATION RESULTS AND ANALYSIS
4.1 SENSITIVITY TO NETWORK SIZE
The network has low mobility , where Vmax is
1 meter per second (m=s), and low transmission error rate
(Perr = 1%). The data traffic load CPR is 10 packets .
4.2 SIMULATION TOPOLOGY
Fig 4.1.1 Link between Source and Destination node
The Fig.6.1.1 shows the simulation topology. Here
heterogeneous network is considered. The 5 source nodes
are connected to the router by means of wired connection.
Similarly the receiver nodes are connected to means of the
wired connection. The connection between the routers is
wireless. The bursty traffic is considered to analyze the
congestion effects.
4.3 SIMULATIONRESULTS
International Research Journal of Engineering and Technology (IRJET) e-ISSN: 2395-0056
Volume: 06 Issue: 03 | Mar 2019 www.irjet.net p-ISSN: 2395-0072
© 2019, IRJET | Impact Factor value: 7.211 | ISO 9001:2008 Certified Journal | Page 7567
5.CONCLUSION
Available bandwidth measures the quantity of the
maximum throughput of a link. It is used to transmit data
without disrupting any existing flows in the networks. The
inaccurate estimation of the bandwidth of the link in
wireless multi-hop networks leads to wireless networks
performance degradation. For lossy MANETs, duplicate
packets were transmitted on multiple routes to enhance
reliability, which consumed extra bandwidth. The
proposed multicast protocol can avoid the HRP/ HMRP
while providing bandwidth guarantees to a requested flow
and ongoing flows. There is no redundant packet
generated by this multicast protocol as a consequence of
using the randomized network coding.
6.REFERENCES
1. Afanasyev, A, Tilley. N, Reiher.P and Kleinrock. L
(2010), “Host-to-Host Congestion Control for
TCP”, IEEE Communications Surveys and
Tutorials, Vol. 12, No.3, pp.1-45.
2. Brakmo.L and Peterson.L (2001), “TCP Vegas: End
to End Congestion Avoidance on a Global
Internet”, IEEE Journal on Selected Areas in
Communication, Vol.13, No.8, pp. 1465-1480.
3. Casetti.C (2002) , “TCP Westwood: Bandwidth
estimation for enhanced, transport over wireless
links”, 7th International Conference on Mobile
computing and networking ,Vol. 8, No.3, pp. 467-
469
4. Hung-Yun Hsieh, Kyu-Han Kim, Yujie Zhu, and
Raghupathy Sivakumar (2006), “A Receiver-
Centric Transport Protocol for Mobile Hosts with
Heterogeneous Wireless Interfaces”, IEEE/ACM
Transactions on Networking, Vol. 11, pp.363-382.
5. Srinivas.K and Dr.Chari.A.A (2010), “An updated
congestion control algorithm for TCP throughput
improvement in wired and wireless networks”,
IEEE journal on selected areas in communication,
Vol.9, No.2, pp. 235-248.
6. Tan.K and Song.J (2007), “A compound TCP
approaches for high-speed and long distance
networks”, 25th IEEE International Conference on
Computer Communications,Vol.11, pp.1-12.
7. Wei.D.X, Jin.C, Low.S.H, and Hedge.S (2006), “FAST
TCP: Motivation, architecture, algorithm and
performance”, IEEE/ACM Transactions on
Networking , Vol. 14, No. 6, pp.1246 – 1259.
8. Xin Ming Zhang, Wen Bo Zhu, Dan Keun Sung
(2010), “TCP Congestion Window Adaptation
Through Contention Detection in Ad Hoc
Networks”, IEEE Transactions on Vehicular
Technology, Vol. 59, No. 9,pp.4578-4588.
9. Xu.L, Harfoush.K and Rhee. I (2006), “Binary
Increase Congestion Control (BIC) for Fast Long
Distance Networks”, IEEE INFOCOM, Vol.4,
pp. 2514 – 2524.
10. Yu-Husan Chen and Eric Hsiao-Kuang (2017),
“Bandwidth-Satisfied and Coding-Aware Multicast
Protocol in MANETs”, IEEE Transactions on
Mobile Computing, Vol. 11, No. 2,pp.1-13.

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IRJET- QOS Based Bandwidth Satisfaction for Multicast Network Coding in Manet

  • 1. International Research Journal of Engineering and Technology (IRJET) e-ISSN: 2395-0056 Volume: 06 Issue: 03 | Mar 2019 www.irjet.net p-ISSN: 2395-0072 © 2019, IRJET | Impact Factor value: 7.211 | ISO 9001:2008 Certified Journal | Page 7563 QOS BASED BANDWIDTH SATISFACTION FOR MULTICAST NETWORK CODING IN MANET C.Shalini1 , K.Senthil Prakash2 1. Second Year-M.E. (Applied Electronics), Velalar College of Engineering and Technology, Erode. 2. Assistant Professor(Sl.Gr), Department of ECE, Velalar College of Engineering and Technology, Erode. -----------------------------------------------------------------------------***--------------------------------------------------------------------------- ABSTRACT-Wireless links are lossy due to network traffic and fading channel. To avoid the packet loss in this method duplicate packet is forwarded in the link to get the acknowledgement from a receiver. Multipath routing trees are constructed for the transmission which providing the bandwidth guarantee. With the randomized network coding, the probability of overhearing the same packet transmission by different forwarder is very low. The proposed multicast routing protocol is designed based on contention-based media access control (MAC) protocols. Scheduling algorithm is not separately needed for distributing packets among the constructed trees. Estimation of the bandwidth before transmission is obtained using a variable bit rate. To improve the accuracy of system and earlier estimation of the bandwidth, multicast routing protocols are used. This method reduced the delay and retransmission rates and improve the bandwidth usage in the MANET. Key Words: MANET, QOS, HRP, HMRP. 1. INTRODUCTION A mobile ad hoc network (MANET) enables wireless communications between participating mobile nodes without the assistance of any base station. Two nodes that are out of one another’s transmission range to need the support of intermediate nodes, which a relay messages to set up a communication between each other. The most fundamental role in MANET is the broadcast operation because of the broadcasting nature of radio transmission: When a sender transmits a packet, all nodes within the sender’s transmission range will be affected by this transmission. The advantage is if one node transmits a packet, all its neighbors can receive this message. In the negative side, one transmission may interfere with other transmissions, creating the exposed terminal problem where an outgoing transmission collides with an incoming transmission and the hidden terminal problem where two incoming transmissions collide with each other. Some existing QoS-guaranteed multicast protocols for MANETs assume ideal links. However, links are lossy in real wireless networks. The packet loss ratio, which represents the probability that a packet is lost or erroneous, depends on the channel quality. When the packet loss ratio of a lossy link increases, more bandwidth consumption for retransmission is required .Since real wireless links are lossy, it is practically important to consider lossy MANETs when we are designing a bandwidth-satisfied multicast protocol. It is a great challenge to design a multicast routing protocol for lossy MANETs that can provide bandwidth guarantees, avoid redundant packets, and reduce the bandwidth consumption at the same time. Providing bandwidth guarantees to traffic flows may incur lots of redundant packet due to lost packets. No bandwidth-satisfied unicast/multicast routing protocol for lossy MANETs was proposed MANETs suffer from a high transmission error rate because of the high transmission contention and congestion. Furthermore, it is a major challenge to provide high reliability for broadcasting operations under such dynamic MANETs. 2. EXISTING SYSTEM In RACC mechanism, the receiver not only performs the function of flow control, but also participates in the congestion control. It first measures the bandwidth, and then computes an appropriate congestion window size based on the measured bandwidth and the RTT. To perform these functions, the receiver has to maintain two timers: one timer for recording the packet inter-arrival interval and the other for measuring the RTT. The sender makes use of this information about its receiver to adjust the congestion window (Xu 2006). 2.1 FUNCTION OF RECEIVER The bandwidth is measured from the receiver according to the packet inter-arrival interval. This method
  • 2. International Research Journal of Engineering and Technology (IRJET) e-ISSN: 2395-0056 Volume: 06 Issue: 03 | Mar 2019 www.irjet.net p-ISSN: 2395-0072 © 2019, IRJET | Impact Factor value: 7.211 | ISO 9001:2008 Certified Journal | Page 7564 can remedy the oscillation in the estimation of bandwidth of TCP Westwood (Casetti 2002). Let Bw be the measured bandwidth, L be the data packet size, and tint be the packet inter-arrival interval. Then, one can estimate the available bandwidth by Bw =L/tint for each packet arrival. Moving average method is used within each congestion window. Let Bw be the i-th measured bandwidth. Then, the bandwidth can be continuously updated by below the equation (3.1). Bw=αBw+(1-α)Bwi (3.1) Where is an exponential filter co-efficient, =0.9 is a good value because the former averaged values should have a higher weight (0.9 in this mechanism) to lower the measure variations. To reduce the consequence of cross traffic, a low pass filter can also be adopted. The receiver measures the RTT based on the data packet arrival time. A variable rcv.rtt is used to record the RTT. The algorithm can be described as follows. i) When an ACK is sent, if rcv.rtt is zero, let rcv.rtt equal 1 and record the corresponding sequence (rtseq) of data packets (equals to the sum of the ACK sequence and the current congestion window). Otherwise, just send the ACK. ii) If a data packet of a larger sequence number than “rtseq” is arrived in order and the new measured packet inter-arrival interval is larger than two times of pre- estimated packet inter-arrival intervals, set the new measured RTT to the value of rcv.rtt, and let rcv.rtt be zero. iii) On each clock cycle, if rcv.rtt is not zero, rcv.rtt is added by 1. Then, the retransmission timer is setting based on the measured RTT. Next, the receiver uses RTT to convert the bandwidth to the receiver congestion window value rwnd using the below equation (3.2) rwnd=Bw*RTT (3.2) Then, it analyses the rwnd with the available receiver buffer, and deposits the lesser value of the advertised window field of an ACK going back to the sender. It is shown in the below equation (3.3) adv_wnd=min(r_abuf,rwnd) (3.3) In other words, the receiver advertised window not only has the original flow control function, yet also takes on the congestion control function. When the receiver detects a timeout, it will send an ACK to and inform the sender that the network is congested. In this ACK, the receiver advertised window is set to one packet, meaning that the sender must retransmit the dropped packet. As the receiver must have detected this timeout earlier than the sender, it will help the sender to reduce the waiting time and confirm the packet loss. 2.1.1 MANAGING THE RTO TIMER An implementation should manage the retransmission timer(s) in such a way that a segment is never retransmitted too early, i.e. less than one RTO after the previous transmission of that segment. The following algorithm is recommended for managing the retransmission timer: 1. Every time a packet containing data is sent, if the timer is not running, start it running so that it will expire after RTO seconds (for the current value of the RTO). 2. When much outstanding data has been acknowledged, turn off the retransmission timer. 3. When an ACK is received that recognizes new data, restart the retransmission timer so that it will expire after RTO seconds. When the retransmission timer expires, do the following: 4. Retransmits the earliest segment that has not been acknowledged by the TCP receiver. 5. The host must set RTO <- RTO * 2 ("back off the timer"). The maximum value may be used to provide an upper bound to this doubling operation. 6. Start the retransmission timer, such that it expires after RTO seconds (for the value of RTO after the doubling operation. The TCP implementation may clear SRTT and RTTVAR after backing off the timer multiple times as it is likely that the current SRTT and RTTVAR are bogus in this situation. Once SRTT and RTTVAR are cleared they should be initialized with the next RTT sample taken. 2.2 REACTION OF SENDER As the receiver can timely help the sender to increase the congestion window according to the instantaneous available bandwidth, the sender only needs to maintain the AIMD mechanism in the congestion avoidance stage, and the slow start stage can be eliminated. Upon a timeout event (whether it is detected by the sender’s timer or informed by the receiver’s ACK), the sender will decrease the congestion window to one in consideration that the network is in congestion (Srinivas 2010) and it will have some time to recover. If congestion is mitigated after one RTT, the sender will recover to adjust the congestion window in the next window by using the receiver advertised window. During fast retransmission, the sender sets the congestion window size of the lesser value of the receiver advertises window size and the current size. Since the packet loss may also indicate congestion, we should reduce the congestion  
  • 3. International Research Journal of Engineering and Technology (IRJET) e-ISSN: 2395-0056 Volume: 06 Issue: 03 | Mar 2019 www.irjet.net p-ISSN: 2395-0072 © 2019, IRJET | Impact Factor value: 7.211 | ISO 9001:2008 Certified Journal | Page 7565 window. On the other hand, if the congestion window is less than the receiver advertised window, the network may only be in a mild congestion (Afanasyev 2010). Therefore, it is unnecessary for the sender to reduce the congestion window. In a normal state, when the sender receives an ACK, it will compare the current congestion window with the receiver advertised congestion window. If the receiver advertised window is larger than the sender’s congestion window, and the difference is larger than a predefined threshold, the sender will set the congestion window size to the receiver advertised window. Or else, the sender will ignore the receiver advertised congestion window by just performing the additive increase mechanism. The threshold value of TCP Vegas (Brakmo 2001) is used. 3.METHODOLOGY A multicast routing protocol for lossy MANETs is proposed. By means of constructing a multiple multicast trees (trees for short) and transmitting coded packets of randomized network coding the proposed multicast routing protocol can provide bandwidth guarantees, avoid redundant packets, and reduce the total bandwidth consumption to a certain degree. With the randomized network coding, the probability that two forwarders overhearing the same packets transmit the same coded packets is very low. The proposed multicast routing protocol is designed based on the contention-based media access control (MAC) protocols. Scheduling algorithm is not separately needed for distributing packets of the constructed trees. In order to provide bandwidth guarantees to multicast destinations, the bandwidth that each source-to-destination route can provide must be aware. The forwarders can provide the minimum bandwidth for the route. To determine the bandwidth that a forwarder can provide, its residual bandwidth and bandwidth consumption are estimated by the proposed multicast routing protocol. Moreover, its one-hop and two- hop neighbors are also examined in order to prevent the hidden route problem (HRP) and the hidden multicast route problem (HMRP) 3.1 NEW BANDWIDTH-SATISFIED MULTICAST ROUTING PROTOCOL In this section, we intend to propose a multicast routing protocol for lossy MANETs that can reduce the total bandwidth consumption to a certain degree while providing bandwidth guarantees to a requested flow and ongoing flows. Multiple multicast trees are constructed so that the residual bandwidth can be fully utilized and the bandwidth consumption can be reduced. In addition, the randomized network coding is applied so that redundant packets can be avoided and each destination can receive innovative coded packets of distinct routes in the proposed multicast routing protocol. A tree construction algorithm is proposed to construct a multicast tree at a time. It selects hosts of better channel conditions as forwarders to reduce the total bandwidth consumption. Each constructed tree connects some destinations, and each can provide a predefined percentage of the bandwidth requirement of each connected destination. The tree construction will continue until all destinations are bandwidth satisfied. There is a basic procedure contained in the proposed algorithm. 3.2 TREE CONSTRUCTION ALGORITHM Given a requested flow Γ, the tree construction algorithm can construct one or more trees. Sequentially, to provide bandwidth guarantees to the destinations of Γ and to reduce the total bandwidth consumption. There are four input parameters, i.e., hs, D, b ¨ _req, and b_per, for the algorithm, where hs is the source, D¨ is the set of all destinations of Γ, b_req is the bandwidth requirement of each destination, and b_per is a predefined percentage for the bandwidth requirement (i.e., b_req). There are three sets D, D , D of destinations, one set F of forwarders, and a series of 2-D arrays B1, B2,...,Bϕ, B˜ used in the algorithm, where ϕ is the number of trees constructed. Given a host hi and a destination hd, Bt [i, d] records the bandwidth of hi that is consumed for hd in the tth constructed tree, where 1 ≤ t ≤ ϕ, whereas B˜[i, d] accumulates the bandwidth of hi that is consumed for hd in all constructed trees. 3.3 ESTIMATION OF AVAILABLE BANDWIDTH The available bandwidth estimation is needed in wireless network because there is no control in the flow of data and gets affected due to the interference such as network traffic, link failures and the channel may either over-utilized or underutilized. Therefore, a deterministic approach to provisioning service generally overestimates the actual resource needs, resulting in a low utilization of network resources. The majority of QOS solutions are loaded with capability for measuring the jitter, delay, available bandwidth. The term available bandwidth can be defined as the maximum throughput that can be transmitted on one link or between two neighbor nodes without influence on any existing flow in the network. The bandwidth reservation solutions designed to 802.11 based
  • 4. International Research Journal of Engineering and Technology (IRJET) e-ISSN: 2395-0056 Volume: 06 Issue: 03 | Mar 2019 www.irjet.net p-ISSN: 2395-0072 © 2019, IRJET | Impact Factor value: 7.211 | ISO 9001:2008 Certified Journal | Page 7566 ad hoc networks use stochastic estimations of available bandwidth. Stochastic bandwidth estimations of random service are based on inferring an unknown bounding function of measurements of variable bit rates (VBR) probing traffic. It is one of the processes to monitor the channel usage based on the CSMA scheme for IEEE 802.11. However, all the existing approaches do not consider the random factors in bandwidth estimations. The proposed system is to improve the accuracy of available bandwidth estimations by considering the random factors such as incoming rates; outgoing rate and packets type consideration (e.g., control packets and data packet). 4.SIMULATION RESULTS AND ANALYSIS 4.1 SENSITIVITY TO NETWORK SIZE The network has low mobility , where Vmax is 1 meter per second (m=s), and low transmission error rate (Perr = 1%). The data traffic load CPR is 10 packets . 4.2 SIMULATION TOPOLOGY Fig 4.1.1 Link between Source and Destination node The Fig.6.1.1 shows the simulation topology. Here heterogeneous network is considered. The 5 source nodes are connected to the router by means of wired connection. Similarly the receiver nodes are connected to means of the wired connection. The connection between the routers is wireless. The bursty traffic is considered to analyze the congestion effects. 4.3 SIMULATIONRESULTS
  • 5. International Research Journal of Engineering and Technology (IRJET) e-ISSN: 2395-0056 Volume: 06 Issue: 03 | Mar 2019 www.irjet.net p-ISSN: 2395-0072 © 2019, IRJET | Impact Factor value: 7.211 | ISO 9001:2008 Certified Journal | Page 7567 5.CONCLUSION Available bandwidth measures the quantity of the maximum throughput of a link. It is used to transmit data without disrupting any existing flows in the networks. The inaccurate estimation of the bandwidth of the link in wireless multi-hop networks leads to wireless networks performance degradation. For lossy MANETs, duplicate packets were transmitted on multiple routes to enhance reliability, which consumed extra bandwidth. The proposed multicast protocol can avoid the HRP/ HMRP while providing bandwidth guarantees to a requested flow and ongoing flows. There is no redundant packet generated by this multicast protocol as a consequence of using the randomized network coding. 6.REFERENCES 1. Afanasyev, A, Tilley. N, Reiher.P and Kleinrock. L (2010), “Host-to-Host Congestion Control for TCP”, IEEE Communications Surveys and Tutorials, Vol. 12, No.3, pp.1-45. 2. Brakmo.L and Peterson.L (2001), “TCP Vegas: End to End Congestion Avoidance on a Global Internet”, IEEE Journal on Selected Areas in Communication, Vol.13, No.8, pp. 1465-1480. 3. Casetti.C (2002) , “TCP Westwood: Bandwidth estimation for enhanced, transport over wireless links”, 7th International Conference on Mobile computing and networking ,Vol. 8, No.3, pp. 467- 469 4. Hung-Yun Hsieh, Kyu-Han Kim, Yujie Zhu, and Raghupathy Sivakumar (2006), “A Receiver- Centric Transport Protocol for Mobile Hosts with Heterogeneous Wireless Interfaces”, IEEE/ACM Transactions on Networking, Vol. 11, pp.363-382. 5. Srinivas.K and Dr.Chari.A.A (2010), “An updated congestion control algorithm for TCP throughput improvement in wired and wireless networks”, IEEE journal on selected areas in communication, Vol.9, No.2, pp. 235-248. 6. Tan.K and Song.J (2007), “A compound TCP approaches for high-speed and long distance networks”, 25th IEEE International Conference on Computer Communications,Vol.11, pp.1-12. 7. Wei.D.X, Jin.C, Low.S.H, and Hedge.S (2006), “FAST TCP: Motivation, architecture, algorithm and performance”, IEEE/ACM Transactions on Networking , Vol. 14, No. 6, pp.1246 – 1259. 8. Xin Ming Zhang, Wen Bo Zhu, Dan Keun Sung (2010), “TCP Congestion Window Adaptation Through Contention Detection in Ad Hoc Networks”, IEEE Transactions on Vehicular Technology, Vol. 59, No. 9,pp.4578-4588. 9. Xu.L, Harfoush.K and Rhee. I (2006), “Binary Increase Congestion Control (BIC) for Fast Long Distance Networks”, IEEE INFOCOM, Vol.4, pp. 2514 – 2524. 10. Yu-Husan Chen and Eric Hsiao-Kuang (2017), “Bandwidth-Satisfied and Coding-Aware Multicast Protocol in MANETs”, IEEE Transactions on Mobile Computing, Vol. 11, No. 2,pp.1-13.