International Journal of Emerging Trends & Technology in Computer Science (IJETTCS)

Web Site: www.ijettcs.org Email: editor@ijettcs.org

Volume 4, Issue 1, January-February 2015
ISSN 2278-6856

Video Streaming: Challenges to Innovation

Mr. R B Mete1, Prof. A G Korke2
1

Solapur University, SVERIS College of Engineering,

Pandharpur, Maharashtra, India

Solapur University, SVERIS College of Engineering,

Pandharpur, Maharashtra, India

Abstract
With the tremendous use of internet throughout the whole
world, multimedia services provided are widely increased. And
it will again increase in future. Transmission of Multimedia
over the internet require the quality in all respect .But till
wireless channels are more unreliable as compared to wired
channels & bandwidth of these wireless channels changes
with time. Due to different types of receivers, it is also difficult
to obtain efficiency for video multicast. This paper discusses
the sender-to-receiver Quality of Service provisioning for
scalable video streaming traffic delivery over existing method.
The Prototype architecture is proposed & validated which
explores the joint use of packet prioritization and scalable
video coding (SVC) together with the appropriate mapping of
UMTS traffic classes to the Different traffic classes. We
develop an analytical model to determine tradeoffs between
network parameters using CSMA MAC protocol. On top of the
analytical model, we use an optimization approach for
determining the optimal number of APs in a cluster and the
optimal separation distances between APs. Several approaches
have been proposed in order to address the end-to-end QoS
both from the network perspective, like UMTS QoS traffic
classes, and from the application perspective, like scalable
video coding and packetized prioritization mechanisms. This
paper is to find the end-to-end QoS problem of scalable video
streaming traffic delivery. We develop an frequency planning
to improve the capacity of network and by eight unique video
sequences and two scalable encoders, demonstrates the
quality gains of scalable video coding(SVC).

Keywords:- SVC, QoS, BER, SAMM, JSCC.

1. INTRODUCTION
Real-time video transmission is largely achieved through
applications that impose an end-to-end delay constraint on
the video stream. Those real-time applications include
conversational applications such as videoconferencing,
distance learning, and video phony. Such applications
usually have strict end-to end delay constraint, less than
200 milliseconds. Real-time applications may also include
streaming applications. Those applications allow the start
of video playback before the whole video stream has been
transmitted with an initial setup time usually of a few
seconds. All those applications require real-time playback.
That is, once the playback starts, it must be continuous
without interruption. Real-time video applications have
gained increased popularity since the introduction of the
first commercial products for Internet video streaming in
1995. As wireless networks are quickly becoming an
important component of the modern communications

Volume 4, Issue 1, January February 2015

infrastructure, Internet protocol (IP)-based architecture for

the fourth-generation (4G) wireless systems[1] grows to be
the provider of the next generation wireless services such
as voice, high-speed data, Internet access, and multimedia
streaming on all IP networks. The high bit rate support in
the WIMAX standard makes it possible to transmit video
in WIMAX [2]. All the above-mentioned cutting-edge
developments, however, confront the high technical
hurdles associated with high bit rate, quality of service
(QoS), and real-time requirements of video applications.
Quality of Service (QoS) means the capability of a
network to provide better service to selected network
traffic over various technologies. For technologies like
Frame Relay, Asynchronous Transfer Mode (ATM),
Ethernet and 802.1 networks, SONET, and IP-routed
networks that may use any or all of these underlying
technologies. The primary goal of QoS is to provide
priority including dedicated bandwidth, controlled jitter
and latency (required by some real-time and interactive
traffic), and improved loss characteristics. Also important
is to make sure that providing priority for one or more
flows does not make other flows fail. QoS technologies
provide the elemental building blocks that will be used for
future business applications in campus, WAN, and service
provider networks. This article outlines the features and
benefits of the QoS provided by the Cisco IOS QoS[].
1.1 Challenges for Real-Time Video Transmission
Generally speaking, the main challenge to the real-time
video communications is how to reliably transmit video
packets over error-prone networks, where meeting the
transmission deadline is complicated by the variability in
throughput, delay, and packet loss in the network. In
particular, a key problem of video transmission over the
existing Internet and wireless networks is the
incompatibility between the nature of the network
conditions and the QoS requirements (such as those
pertaining to bandwidth, delay, and packet loss) of
multimedia applications. With a best-effort approach, the
current IP network was originally designed for data
transmission, having no guarantee of QoS for multimedia
applications. Similarly, the current wireless networks were
designed mainly for voice communication, which does not
require as large bandwidth as video applications do.
Different types of IP applications have different types of
quality impairment under the same network conditions,
and therefore call for different QoS. For example,
applications such as web browsing, data file transfer, and
electronic mail, are not sensitive to delay. However, for
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International Journal of Emerging Trends & Technology in Computer Science (IJETTCS)

Web Site: www.ijettcs.org Email: editor@ijettcs.org
Volume 4, Issue 1, January-February 2015
ISSN 2278-6856
the deployment of multimedia applications with video
stream, which is more sensitive to delay but more tolerant
to packet loss, the lack of QoS guarantees in today's
Internet and wireless networks introduces huge
complications.
Specifically,
several
technological
challenges need to be addressed in designing a highquality video transmission system. First, to achieve
acceptable delivery quality, transmission of a real-time
video stream typically has a minimum bandwidth
requirement. However, the current Internet does not
provide bandwidth reservation to meet the bandwidth
requirement. At the same time, compared to wired links,
wireless channels are much noisier due to fading, multipath, and shadowing effects, which result in a much
higher bit error rate (BER) and consequently an even
lower throughput. Second, in the Internet, a packet can be
lost due to congestion caused by buffer overflow and
excessive delay. In wireless networks, a packet with
unrecoverable bit errors is usually discarded at the linklayer according to the current standards. This difficulty is
not as severe for traditional IP applications such as data
transfer and email, where reliable transmission can always
be achieved through retransmission, as it is for real-time
video applications, where retransmission-based techniques
may not be available due to the tight delay constraints.
Third, the network resources are limited and may vary
with time and space. Network resources include
transmission bandwidth, buffers in the routers and
switches, buffers at the sender or the receiver end,
computation capability for encoding, decoding and
transcoding, transmission cost in networks with pricing
charge enabled, transmission power in wireless
communications, delay, etc. Some constraints on resource,
such as buffer size, computation speed, and display
precision at the user end, but other constraints in that they
aim to make the system stable or to treat other users in the
communication system fairly. One example of a soft
constraint at hand is the TCP-friendly protocol used to
perform congestion control for media delivery
applications, where the source bit rate is constrained so
that all kinds of traffics can fairly share the network
resources.
1.2 Existing System
In our existing system, proposed the Joint source-channel
coding, Similar to error resilient source coding. In
general, JSCC is accomplished by designing the quantize
and entropy coder for given channel. For image and video
signals, JSCC focuses on the optimal bit allocation
between source coding and channel coding ,given channel
loss characteristics. JSCC is studied for wavelet image
transmission over the Internet, where the source and
channel coding bits are allocated to minimize the expected
end-to-end distortion. In JSCC for Internet scalable video
is studied, where error resilient source coding and FEC
are jointly considered[3]. The similar problem of scalable
video transmission is also over bit-based channels. We
take ARQ into account for non-scalable video. Real-time
video applications, such as videoconferencing, video
phony and on-demand video streaming, have gained

Volume 4, Issue 1, January February 2015

increased popularity. However, a key problem of video

transmission over the existing Internet and wireless
networks is the incompatibility between the nature of the
network conditions and the QoS (Qualify of Service)
requirements (such as those in bandwidth, delay, and
packet loss) of real-time video applications. Cross-layer
design is a natural approach to deal with the
incompatibility problem. This approach aims to efficiently
perform cross-layer resource allocation (such as
bandwidth, transmission energy, and buffers) by
increasing the communication efficiency of multiple
network layers.
Demerits:
1)Video quality reduced.
2)Video streaming time is increased.
1.3 Literature Survey
Optimal rate allocation for scalable video multicast over
Wi-MAX, by Hsin-Yu Chi, Chia-Wen Lin and YungChang Chen[4] has been proposed a new wireless
broadband standard, which is capable of delivering very
high data rate and covering wide area. The above said
paper propose a Genetic Algorithm (GA) to reduce
computational complexity when finding optimal
modulation. The proposed method achieves lowest
average distortion and the reduced computational
complexity. According to the size of video layer, SS
distribution, and available symbols; our method adaptively
changes the modulations of each video layer in each group
of picture (GOP) time. To ensure each receiver could
receive all data of BL, we choose the minimum one of the
modulation groups whose SS number is not zero in a
multicast session. As a result, the PER of BL for receiver
should be made zero. After allocating the available
symbols to the data of BL, the remaining symbols are
allocated to ELs according to the order of layer index.
Furthermore, we assume the given number of available
symbols must be larger than the symbols required for
sending the BL data. In our GA-based algorithm, we use
the Roulette wheel selection algorithm to perform fitnessproportional selection according to the fitness value of the
chromosomes. The requirement is that the fittest
individuals have a greater chance of survival than weaker
ones. Besides, we use the one-cut-point crossover method
to obtain offspring with a crossover rate. Based on the
mutation rate, some portion of genes will be changed. To
ensure that the high fitness chromosome is not lost from
generation to generation, the best chromosome is copied
into the new generation.
1.4 A Cross-Layer Design for Scalable Mobile Video
This paper presents SoftCast, a clean-slate design for
mobile video[5]. This paper aims to improve the
robustness and scalability of mobile video. The paper
presents SoftCast, a cross-layer design of mobile video
that both compresses the video and protect it from errors
and losses. SoftCast adopts an integrated design for video
and PHY layer coding, making the whole network stack
act as a linear transform. The lack of scalability and
robustness in todays mobile video stems from the existing
design of the network stack. Specifically, mobile video is
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International Journal of Emerging Trends & Technology in Computer Science (IJETTCS)

Web Site: www.ijettcs.org Email: editor@ijettcs.org
Volume 4, Issue 1, January-February 2015
ISSN 2278-6856
impacted by two layers in the stack: the application video
codec, which compresses the video, and the physical layer,
which protects the video from channel errors and losses.
This paper aims to improve the robustness and scalability
of mobile video. SoftCast starts with a video that is
represented as a sequence of numbers, with each number
representing a pixel luminance. It then performs a
sequence of transformations to obtain the final signal
samples that are transmitted on the channel. The crucial
property of SoftCast is that each transformation is
linear[5]. This ensures that the signal samples transmitted
on the channel are linearly related to the original pixel
values. Therefore, increasing channel noise progressively
perturbs the transmitted bits in proportion to their
significance to the video application; high-quality
channels perturb only the least significant bits while lowquality channels still preserve the most significant bits.
Thus, each receiver decodes the received signal into a
video whose quality is proportional to the quality of its
specific instantaneous channel.
1.5
Layered Resource Allocation for Video
Broadcasts over Wireless Networks
This paper aims to combine adaptive modulation and
coding with layered video coding to improve the quality of
video services to users experiencing different radio
conditions[6]. This paper proposes an optimal radio
resource allocation algorithm which maximizes a general
performance metric for a video session in polynomial
time. In this paper, optimal allocation algorithm runs in
polynomial time, making it suitable for practical video
services, and maximizes a general utility function. Also an
alternative formulation of utility can be adapted to meet
different circumstances. In this paper, a new layered
resource allocation is proposed which fully utilizes the
AMC and layered coding facilities in broadcast standards.
With Adapting a generalized performance metric to
accommodate various quality measures, we will describe
an algorithm which maximizes the quality of each video
session, while guaranteeing a maximum video quality to
all users; and this algorithm runs in polynomial time
under any given resource budget. We show that a systemwide optimal resource allocation can be obtained with a
simple two-step decomposition of allocation that deals
with each video session and then with the system as a
whole. We also present a sub-optimal system-wide
allocation algorithm of reduced computational complexity,
which is useful when the distribution of users radio
conditions changes frequently. This work extends existing
broadcast and multicast services by the introduction of the
layered resource allocation, which enables fine-grained
control over coverage and video quality when many users
are experiencing heterogeneous radio conditions. Our
optimal allocation algorithm runs in polynomial time,
making it suitable for practical video services, and
maximizes a general utility function, so that alternative
formulation of utility can be adapted to meet different
circumstances.

Volume 4, Issue 1, January February 2015

1.6
Source-Adaptive Multilayered Multicast
Algorithms for Real-Time Video Distribution
Multilayered encoding, however, is not sufficient to
provide high video quality and high network utilization.
Adaptive techniques capable of adjusting the rates of video
layers are required to maximize video quality and network
utilization. We propose a class of algorithms known as
source-adaptive
multilayered
multicast
(SAMM)
algorithms. SAMM algorithms exhibit good performance
in terms of good put, video quality and scalability while
requiring only a minor amount of buffer allocation[7].
This paper also introduces a network architecture defining
the source, receiver and network functions necessary to
support SAMM algorithms. This architecture includes
feedback mergers, which consolidate the contents of
feedback packets as they return to the source. Feedback
mergers are used to prevent feedback implosion, a
scalability problem that occurs as an increasing number of
receivers return congestion feedback to a single source.
This architecture also requires that routers implement
some form of priority packet discarding, to ensure that, in
case of congestion, packets from less important video
layers are discarded before packets from more important
layers. In order to prevent SAMM video flows from
negatively impacting the performance of other adaptive
flows in the network, routers must also isolate flows based
on service class. If this extraneous traffic shares first-infirst-out (FIFO) queues with competing traffic that is
adaptive (e.g., TCP flows), then the adaptive flows may
experience an unfair degree of discarding or delay within
the network. Receiver-driven algorithms do not share this
deficiency with sender-driven algorithms, because they
send each layer of video in a different flow and allow for
the pruning of flows that have no downstream receivers.
One way to correct this deficiency of the sender-driven
algorithms is to isolate video traffic from other traffic.
1.7
An adaptive modulation method for multicast
communications of hierarchical data in wireless
networks
This paper propose a new adaptive modulation method for
multicast communications. The method separates data
into hierarchical layers according to its importance, and
transmits the layers of data in the same frame
adaptively[8]. The number of slots for each physical layer
mode is optimized, and a simple closed form solution is
obtained. The relative size of each layer within a frame
and the data rates the layers are sent are chosen to
maximize the total sum of data rates for a given outage
probability. The first criterion is the sum of data rates that
are successfully detected at the receiver. The perceived
quality of hierarchical multimedia is proportional to the
number of successfully received layers by each user. Thus,
the sum of data rates can be considered as the sum of
perceived quality of the multicast service. The second
criterion is the outage probability, which is the probability
that a receiver cannot receive the base layer at a required
BER. A low outage probability ensures a high availability,
which is of particular interest to the service providers. It
separates a set of data into several hierarchical layers
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Web Site: www.ijettcs.org Email: editor@ijettcs.org
Volume 4, Issue 1, January-February 2015
ISSN 2278-6856
according to their importance, and the transmitter sends
the layers of data at different data rates, or modes, in each
data frame. The base layer that contains the most
important information is sent at the lowest data rate while
the higher layers are sent at higher rates. The relative size
of each layer within a frame and the data rates the layers
are sent; are chosen to maximize the total sum of data
rates for a given outage probability. We performed
simulations for 802.11a wireless LAN, which show that
the performance of the proposed method.

2. METHODOLOGY
2.1 Cross-Layer Resource Allocation
We can address the above challenges by enforcing error
control, especially through unequal error protection (UEP)
for video packets that are usually of different importance.
Error control techniques, in general, include error resilient
source coding, forward error correction (FEC),
retransmission, power control, network QoS support, and
error concealment. To maximize the error control
efficiency, limited network resources should be optimally
allocated to video packets which typically requires crosslayer design. The traditional layered protocol stack, where
various protocol layers can only communicate with each
other in a restricted manner, has proved to be inefficient
and inflexible in adapting to the constantly changing
network conditions. For the best end-to-end performance,
multiple protocol layers should be jointly designed and
should be able to react to the channel conditions in order
to make the end-system network-adaptive. In addition,
conventional video communication systems have focused
on video compression, namely, rate-distortion optimized
source coding, without considering other layers. While
these algorithms can produce significant improvements in
source-coding performance, they are inadequate for video
communications over hostile channels. This is because
Shannon's separation theorem, that source coding and
channel coding can be separately designed without any
loss of optimality, does not apply to general time-varying
channels, or to systems with a complexity or delay
constraint. Therefore, recent video coding research has
been focused on the investigation of joint design of endsystem source coding with manipulations in other layers,
such as channel coding, power adaptation in wireless
networks, and QoS support from the network (e.g.,
differentiated services networks and integrated services
networks). One of the main characteristics of video is that
different portions of the bit stream have different
importance in their contribution to the quality of the
reconstructed video. For example, in an MPEG video bit
stream, I (Intra) frames are more important than P
(Predictive) and B (Bi-directional predictive) frames. If
the bit stream is partitioned into packets, Intra-coded
packets are usually more important than Inter-coded
packets. If error concealment is used, the packets that are
hard to conceal are usually more important than easily
concealable ones. In the scalable video bit sstream, the
base layer is more important than the enhancement layer.

Volume 4, Issue 1, January February 2015

Therefore, UEP is naturally preferable in video

transmission. UEP commonly enables a prioritized
protection for video packets through different levels of
FEC and/or retransmission. UEP can also be realized
through prioritized transmission by techniques based on
transmitter power adaptation, channel/path diversity.
Thus, in different network infrastructures, we consider
different applicable error control components to support
UEP. In particular, we jointly consider source coding and
cross layer resource allocation to perform optimal UEP.
This Paper is about the end system design for real-time
video applications. Such end system design consists of
three major components: video codec, rate control or
congestion control, and error control. This paper focuses
on error control. More specifically, our focus is on the
interaction between the video encoder and the underlying
layers. Thus the decoder-based techniques, such as postprocessing, joint source-channel decoding, and receiverdriven channel coding are not discussed in this
dissertation. With respect to cross-layer design at the
sender side, traditionally, there are two approaches to
address the challenges described above for video
communications. In the networking community, the
approach is to develop protocols and mechanisms to adapt
the network to the video applications. One example is to
modify the mechanisms implemented in the
routers/switches to provide QoS support to guarantee
bandwidth, bounded delay, delay jitter, and packet loss for
video applications. Another example is to use additional
components such as an overlay network. The networking
community approach is beyond the scope of this paper. In
the video community, on the other hand, the approach is
to adapt the end system to the network, which is what we
employ in this paper. We focus on the end-system design
for video communications based on the current, recently
proposed, and emerging network protocols and
architectures. We assume that the lower layers provide a
set of given adaptation components; from the encoder's
point of view, these components can be regarded as
network resource allocation. Based on the assumption that
our encoder can access and specify those resource
allocation knobs, the major contribution of this paper is a
proposed general resource-distortion optimization
framework, which assigns network resources from
multiple layers to each video packet according to its level
of importance. Depending on different adaptation
components, this framework is embodied in the forms of
joint source-channel coding, joint source coding and rate
adaptation, joint source-channel coding and power
adaptation, joint source coding and packet classification
etc. Within the framework, error resilient source coding,
channel coding, and error concealment are jointly
considered. In particular, we study the following problems
using the proposed framework: joint source-channel
coding (JSCC) for Internet video transmission, joint
source-channel coding and power adaptation (JSCCPA)
for wireless video transmission, and joint resource
allocation and packet classification (JRAPC) for network
video transmission. Besides single layer video source
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Web Site: www.ijettcs.org Email: editor@ijettcs.org
Volume 4, Issue 1, January-February 2015
ISSN 2278-6856
coding, we also consider scalable video source coding for
further error resilience. In addressing these problems, we
propose efficient algorithms for obtaining the optimal
solutions. The simulation results, as expected,
demonstrate the benefits of joint design of source coding
and cross layer resource allocation. The proposed
framework proves to be general and flexible. It allows for
the comparison of different error control techniques (such
as pure FEC, pure retransmission, and hybrid FEC and
selective retransmission), and different packetization
schemes. For example, in the JSCC work, FEC and
application layer retransmission can each achieve an
optimal result depending on the packet loss rates and
roundtrip-time. But when the two are jointly employed in
the proposed hybrid technique, improved results are
obtained due to the increased exibility[9]. The framework
is also applicable to the emerging network architectures,
in which source coding can be jointly designed with
packet classification. The proposed framework provides an
optimization benchmark against which the performances
of other sub7-optimal systems can be evaluated. It also
provides a useful tool for assessing the effectiveness of
different error control components in practical system
design. For example, in the JRAPA study, the simulation
results suggest that channel coding and power adaptation
each has its effective working region. Thus, in a practical
wireless video streaming system, under certain conditions,
adjusting only channel coding or power control, but not
both, might be adequate to achieve near-optimal results.
2.2 Proposed System
In proposed methodology to challenges in QoS
provisioning for wireless video transport, we use an
adaptive framework, which specifically addresses scalable
video transport over wireless networks. This proposed
method to presents a novel method for QoS provisioning
via the use of the average reward reinforcement learning,
which can maximize the network revenue subject to
several predetermined QoS constraints. The proposed
method to investigate the end-to-end QoS provisioning for
scalable video streaming technique. Congestion
management, queue management, link efficiency, and
shaping/policing tools provide QoS within a single
network element. Service levels refer to the actual end-toend QoS capabilities, meaning the capability of a network
to deliver service needed by specific network traffic from
end to end or edge to edge. The services differ in their
level of QoS strictness, which describes how tightly the
service can be bound by specific bandwidth, delay, jitter,
and loss characteristics.
Merits:
a)
he video quality to be increased and improve the
frequency planning to rectify the collision issue for the
video processing in the WMN.
b)
o improve the management of QoS.

Volume 4, Issue 1, January February 2015

Flow Diagram:

Figure 1. Flow Diagram

As shown in figure above, Read video block load the video
files. Video files are converted into frames at video to
frame conversion block. Conversion frame stored in buffer
because to avoid data loss & frames convert into packet in
transmission coder block. Packets are separated for data
transmission. We Transmit the data with the help of
WIMAX protocol. Then receiver receives the packets &
Rearrange it. Received packets are again stored in buffer
at receiver side. The packets are then converted into
frames & the frames are converted into video. Finally
write the video in receiver system. We used Matlab video
processing to develop an application. Matlab provides its
own functionality for video & image processing[10].
2.2.1 Proposed Modules:
Input
Video to frame conversion
Adaptive modulation/demodulation
Transmitter/receiver
Frame to video conversion
QOS constraints
Input
In this module, analysis of the input video format and file
size is done. Some applications create collections of
images related by time, such as frames in a movie, or by
(spatial location, such as magnetic resonance imaging
(MRI) slices). These collections of images are referred to
by a variety of names, such as image sequences, image
stacks, or videos. The toolbox represents image sequences
as four-dimensional arrays, where each separate image is
called a frame, all frames are the same size, and the
frames are concatenated along the fourth dimension.
mmread function will be used for loading and showing the
T
input video sequence.
Video to frame conversion
The loaded video sequence should be converted into
frames (i.e. still images) using mmreader function. To
read the videoTinformation, we are using the read function
which reads the total information about the given video
file. It convert multiple number of frames.

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Volume 4, Issue 1, January-February 2015
ISSN 2278-6856
Adaptive modulation/demodulation
Group the incoming data bits into codewords, one for
each symbol that will be transmitted. Map the codewords
to attributes, for example amplitudes of the I and Q
signals (the equivalent low pass signal), or frequency or
phase values. Adapt pulse shaping or some other filtering
to limit the bandwidth and form the spectrum of the
equivalent low pass signal, typically using digital signal
processing. Perform digital to analog conversion (DAC)
of the I and Q signals (since today all of the above is
normally achieved using digital signal processing, DSP).
Generate a high frequency sine carrier waveform, and
perhaps also a cosine quadrature component. Carry out the
modulation, for example by multiplying the sine and
cosine waveform with the I and Q signals, resulting in the
equivalent low pass signal being frequency shifted to the
modulated pass band signal or RF signal.
Transmitter/receiver
This module contain the data transmitter and receiver
blocks in wimax.
It required receiver antenna and transmitter antenna.
It transmit the data in analog waveform.
It required the IP address for communication process.
Frames to video conversion
The loaded image sequence/frames should be converted
into video (i.e. still images) using video reader function.
To read the image information, we are using the read
function which reads the total information about the given
image file. It converts the multiple number of frames into
video.

Figure 2. Activity Diagram

Input video is converted to frames & then stored in txion
buffer in coded format. Coded frame packets are separated
& transmitted to client. Client decodes received packets &
store in txion buffer. The frames from packet are
converted to video & write to client.

QOS constraints
Real-time video transmission typically has requirements
on quality of service (QoS). However, wireless channels
are unreliable and the channel bandwidth varies with
time, which may cause severe degradation to video
quality. In addition, for video multicast, the heterogeneity
of receivers makes it difficult to achieve efficiency and
exibility.
2.2.2 System Requirement
Hardware Requirement(Minimum):
Pentium IV 2.7 GHz
1GB DDR RAM
250Gb Hard Disk
Software Requirement:
Operating System : Windows XP or win7
Tool : Matlab Version : r2013a

3. RESULT
From the above proposed method, we will get a best
quality video in following way:

Volume 4, Issue 1, January February 2015

Figure 3. BER Plot of sample

BER plot shows BER vs SNR ratio of final video reached
to destination in their respects.
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Volume 4, Issue 1, January-February 2015
ISSN 2278-6856

Figure 4. Throughput of sample

Throughput in Mbps with respect to SNR in db. It shows
Noise rate present in received video & throughput. Here
throughput must be high & noise ratio must be very low.
This graph shows us our received video quality.

Figure 7. Received Signals of sample

Received signals are signals received at client for video
transmission & this graph show signal quality in terms of
AMP with respect to time in seconds. If received signals
are as same as input/transmitted signals then we can say
video is delivered/streamed in best quality. Difference in
input signals & received signals is tolerable for real time
video streaming. Here we have achieved our best quality
of video.

Figure 5. SNR Plot of sample

Figure 8. Server Side/Sending Side Window of sample

This is video sending window. Here video is loaded &
converted to frames. The number of frames created are
shown with the effect of Convert button. Then we have to
connect no. of client to whom we want to connect.

Figure 6. Input Signals of sample

This graph is display at sending side/server side. It shows
input data signals with respect to time in seconds.
Figure 9. Client Side/User Side of sample

Volume 4, Issue 1, January February 2015

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Volume 4, Issue 1, January-February 2015
ISSN 2278-6856
At client side, when we click on Read button ,client
receives frames from sender & convert it to video. This
window also shows received signals. All the above graphs
are changing as input video is changed & as per quality of
video. For example, We have taken a .avi video of 4MB
size & 3 seconds duration. When we have loaded it &
converted then that video is converted in to 54 frames.
The number of frames varies with the size of video. After
a connection with client & client read the video; frames
are transmitted & video display on screen. This
transmission of frames & their conversion into video is
happening in very fast speed i.e. in fraction of seconds.
Also we have investigated different graph plots of
network parameters. These network parameters/graph are
different for different videos.

4. DISCUSION/ ANALYSIS
This technique is useful in live video streaming
applications like video conferencing (i.e. skype
application), to watch video on youtube etc. It can be used
in live traffic monitoring system ,web cam chatting,
CCTV for security. Like this, Now a days many more
applications are used where live video streaming is
required. In this applications, we can use this technique to
improve video quality with performance of application.
Future scope : At Live Video Streaming: This technique
is also very useful where we want to make an artificial
intelligent robot. As we know, In a future, we can use an
automatic robots for many uses of human being. We can
use them for war, we can use them to work in very
hygienic conditions like as worker in dangerous mines,
we can use them for automatically find a suspect & to kill
them(i.e. terrorist) etc. So in these fields of work, we can
add artificial intelligence technique in them but to find a
target & identify it automatically ; we have to load the
image/video from robots front view through the camera
situated in robots eyes, scan that image/video
automatically, identify fixed target & then shoot it. All
this activity should be happen in a fraction of second. In
this mission/activity of robot, video recording , video
storage, live video streaming to server from where robot is
going to operate or where members of that particular
mission are locate, is first and very important task. So,
this technique will be useful in such respect. It is an one
example. This technique has many more real time uses.

5. CONCLUSION
In this paper, We have investigated various factors
affecting live video streaming. These are external &
internal noise, SNR, BER, signal behavior, frequency,
quality etc. We have investigated challenges in real time
video transmission & checked for its solutions like cross
layer resource allocation, Layered Resource Allocation,
Source-Adaptive Multilayered Multicast Algorithms, An
adaptive
modulation
method
for
multicast
communications. For the scalability with the QoS

Volume 4, Issue 1, January February 2015

requirement, an optimization approach has been proposed

to maximize the ratio of total capacity to total cost .We
have developed a model with frequency planning & QoS
constraints. We have used modulation /demodulation
technique for signal transmission & video to frame ,frame
to video conversion technique . The problem of video
leakage in conferencing/streaming is removed with video
compression. We got a fantastic result which is shown
above with graphs ,as an example. We conclude that, this
technique is very effective & also have future scope.

REFERENCES
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AUTHOR
Raviraj Mete received the B.E in
Information Technology from shivaji
university, Maharashtra (2011) &
pursuing ME in computer science at
solapur university, Maharashtra(2015).
Assistant Professor at shri vithal college
of engineering, pandharpur, MH(India).
Ashok Korke
received the B.E in
computer science from solapur university,
Maharashtra & MTech in computer from
walchand college of engineering, sangli,
Maharashtra. Assistant professor at shri
vithal college of engineering, pandharpur, MH(India).

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