Abstract These small but powerful devices are



The conceptual report discusses the project to simulate a video
over IP network with built in QoS features that enables the monitoring of the
performance of the network. The report discusses the current protocols that are
used in Video Over IP, QoS parameters in the transmission of video over IP,
current implementations that facilitate the transmission of video over IP,
benefits and drawbacks of Video Over IP transmission and an overview of the
proposed artefact.

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Chapter 1: Introduction
Chapter 2: Literature Review
2.1 Current
Protocols used for VoIP
2.2 QoS in VoIP
2.4 Benefits of Voice
over IP
Chapter 3: Project
3.1 Project review
3.2 Implementation plan
3.3 Gantt Chart
Chapter 4: Overview
of Artefact
4.1 Requirements
4.2 Design
4.3 Testing Strategy










1:  Introduction

Multimedia applications proliferate the
Internet today; major names such as Youtube and Facebook offer users the
ability to view stored and live multimedia such as video and audio over the
Internet. This explosion in the use of multimedia has been enabled by a similar
rise in the use of smartphones. These small but powerful devices are able to
connect to high speed networks and receive and transmit large amounts of data,
which are required for multimedia transmission. These new multimedia
applications such as Skype, which provides telephony and video conferencing
over the Internet, have resulted in an overhauling of the traditional circuit
based equipment in the Private Branch Exchanges (PBXs), with new soft IP
switches such as Asterisk taking its place. Traditional telephone providers
have also started providing enhanced television services using the existing DSL
(Digital Subscriber Line).

There is however a problem in that the
Internet and related protocols used for data transmission provide only limited
support for multimedia applications. This work explores the constraints faced
in developing an IP (Internet Protocol) network to support high quality video


2: Literature Review

2.1 Current Protocols used in Voice over IP

There are a number of internet related
protocols that are used for the transmission of video over the Internet. For
example FTP (File Transfer Protocol) can be used in the transmission of
video/multimedia files. Peer to peer technology can also be used, and
applications such as BitTorrent provide the interface for peer to peer sharing
of multimedia files. However these methods require the entire file to be
downloaded, and this can require significant amounts of storage. This becomes a
problem particularly when the client devices are smartphones where there may be
limited storage. It is also unsuitable when the video needs to be live;
downloading large video files can be very time consuming, requiring the user to
have to wait for minutes or even hours for the file to be downloaded. Hence
although the downloading and viewing of multimedia files is a robust method of
data sharing it is not sufficiently flexible for most practical multimedia

Streaming is an alternative to downloading.
Streaming is where the media bitstream is split into a number of different
chunks which are transmitted independently over the network; these chunks are
called packets. The receiving node expects the individual packets and has the
ability to put them back together in order to display the video seamlessly to
the end user. The transmitter continues to send the chunks of multimedia data
and the receiver continues to decode, put together and play back the video,
resulting in a quick and constant playback of the video at the user’s end
without delay. This technology which can offer video transmission with low
delay makes it possible for services such as video conferencing and live video
telecast to be offered over the Internet.

Multimedia transmission can be divided into
three types, unicast, multicast and broadcast. As the name suggests, unicast is
a one-to-one communication where one sender sends the multimedia transmission
to one receiver. A video call between two people falls under this category, as
does streaming media on demand. In unicast applications, a feedback channel can
be established between the receiver and transmitter, allowing the receiver to
return information to the sender about channel conditions, end user
requirements, end device characteristics, etc. This in turn allows the sender
to adapt the compression, error protection, and other variables related to the
transmission. Multicast applications allow for the sender to send multimedia
data to multiple receivers; it is more efficient compared to unicasts in terms
of network resource utilization and server complexity. However the sender
cannot target the transmission to the desired recipients, unlike unicast.
Broadcast is where the transmission is sent to all the nodes that are reachable
in the network.

Various different protocols are used in the
transmission of Video over IP today, such as UDP, TCP, RTP and RTCP. Wu et al
(2001) explain that UDP and TCP support multiplexing, error control, congestion
control, etc. Error control for example is implemented by TCP and to a certain
extent, UDP, using checksums to detect bit errors. RTP (Realtime Transport
Protocol) is designed to provide end-to-end transport functions for supporting
real-time applications. RTCP is a companion protocol with RTP and is designed
to provide QoS feedback to the participants of an RTP session. RTP is useful
for the transmission of Video Over IP because it provides for time stamping of
packets to facilitate synchronization of media streams, sequence numbering to
help in reordering packets at the destination and to detect packet losses,
payload type identification, source identification, etc.

2.2 QoS in Voice over IP

D’Orlando et al (n.d.) explain that the QoS
(Quality of Service) available in current networks is insufficient to guarantee
error free delivery of the transmission to all recipients for real time video
transmission. One class of solution is to use video codecs to overcome these
limitations; this works at the application end. Another method is to apply
controls on the network, such as using appropriate protocols. In this work we
only consider adaptation of the network, i.e. changing the network characteristics
to improve the media transmission.

Klaue et al (2003) opine that the following
parameters are relevant to the quality of video transmission over a network:

1.       Delay
and jitter – delay in the arrival of frames and variation of the delay, called
frame jitter, also has a negative impact on the quality of the video
transmission. Digital videos require that the frames are displayed at a
consistent rate. If this is not possible due to the delay in the arrival of the
frames this causes jerkiness of the video. Buffers can be used to absorb the
delays and jitter, because they store the packets that have arrived ready to be
played when necessary. Big enough buffers can store the entire video before it
is played, eliminating all jerkiness due to delay and jitter, this would
however result in a delay in playing the video. Optimisation techniques have
been developed to determine the optimal play out buffers, i.e. the amount of
buffering that would be optimal in a given network in the given network


Brydia et al (2009) suggest that the
following five parameters should be measured and monitored to ensure the
quality of transport of Video over IP:

Inter-packet arrival jitter causing delay

Inter-packet arrival jitter causing burst

Ethernet packet loss

Ethernet inter-packet arrival average
drift/deviation from the Moving Pictures Experts Group (MPEG)-2 data transport

MPEG-2 quality due to packet corruption on the
network, MPEG-2 encoding errors, or MPEG-2 packet loss

Ahmed et al (2009) explain that multimedia
applications have the capability to adapt to different bandwidths, however they
have stringent delay, jitter and packet loss constraints. These are the three
requirements which are also not very well supported by IP networks today.
However they suggest that current QoS models are insufficient because they
operate on per-IP domain, and not on an end to end basis. Service level
agreements (SLAs) are used, but these are hard to implement.

2.3 Current Implementations

There have been numerous solutions that
have been put forward in current literature to address the QoS issues with the
transmission of Video over IP. Ahmed et al (2009) for example put forward a
comprehensive framework which has a few layers – a cognitive layer which is
media and delivery aware; it is aware of the media that is being transported
and the QoS capabilities of the network. This cognitive transport protocol will
be able to sense its surroundings, i.e. identify the service requirements and
underlying network conditions, identify the type of media being delivered, etc.
and then use this information to adapt the network, without any intervention
from the user, such as that the transmission QoS is maximized.
This cognitive transport protocol therefore is essentially an adaptable on the
fly traffic control which is based on changing QoS requirements of the service.

Servetto and Nahrstedt,(2001) also suggest
that a network only view of the QoS problems also has a number of
disadvantages; while it is able to address issues that impact on flow control,
making the resulting video flows very good, this approach still leaves the QoS
at the mercy of the nature of the coders used. Video coders still use
multiresolution techniques and therefore are inherently mismatched to networks
that do not offer packet differentiation. Furthermore, the modified protocols do
no offer error free transmission, which means that at some point the errors
will have a negative impact on the quality of the video transmitted. Therefore
they put forward their solution of a middleware which addresses the mismatch
between the properties of the video coders and properties of the existing
transport protocols. This middleware adjusts the key parameters of the
error-resilient coder as a function of the states of the channel.

2.4 Benefits of Voice over IP

The benefits of Video over IP transmission
are numerous; IP networks today are almost ubiquitous. Wired and wireless local
area networks are often IP networks. Larger networks, such as the Internet
itself, are IP networks. The ubiquity of these networks mean that firstly,
using these networks for video transmission allows for a wide range of users to
be reached. Secondly, it also means that there is a wealth of applications,
services, and other components that have already been developed. This makes it
easier to provide a video over IP service for any entity using these networks.
There is no need to develop most of the required services from scratch (Van de
Schaar and Chou, 2011)

The key cost associated with the
transmission of video over IP is the sacrifice in quality of service. As Wu and
Zhang explain, there is no QoS guarantee for video transmission over the
Internet. Furthermore, heterogeneity of the networks and receivers makes it
difficult to achieve bandwidth efficiency and service flexibility. Winkler and
Mohandas (2008) however caution that the evaluation of QoS of video over IP is
very subjective; Different users may rate the quality of the same video
differently, because different individuals have different interests and
expectations for video. Hence subjectivity and variability of end user rating
of their quality of experience cannot be eliminated.  Therefore any quality score develop will
still be a noisy measurement that is defined by a statistical distribution
rather than an exact number.


Chapter 3: Project Plan

3.1 Project Review

The project will begin with a literature
review, which will explore in detail the specifics of the building of the
network simulations, how network simulations can be useful in the current
project and how they should be tested and evaluated. Then the actual building
of the network simulation will commence. Once the network simulation is
completed then the testing will be done, followed by an evaluation.

3.2 Implementation Plan




create a video over IP network ,
it consists of  we use number hosts 4-10 


implement video calls  pass
through  Video over IP transmission  sender to receiver 
implement RTP and SIP protocol


set the simulation important
parameters are delay=’uniform(20ms,25ms)’,samplingRate =
8000Hz,ompressedBitRate = 40000bps and  datarate=’200kbps’


 implement RTP and SIP
protocol create a module use help of inet framework


Test run the simulation


 plot the results Jitter and
Delay  use omnet vector recording statistics analyses 


3.3 Gantt Chart

Fig 1.
– Project Schedule

3.1 – Gantt Chart

4: Overview of Artefact

4.1 Requirements


The tools that is required in the course of implementation of the
video surveillance system include OMNET+++ and possibly Wireshark to examine
statistics for Jitter and Delay. These are two software that is crucial in the
analysis of various metrics of performance such as Jitter, an end to end Delay
as well as packet movement between one stations to another.


The reason for these statistics is the fact that every system
implementation should be in a position to provide the best results and there
should be few packets lost in between the communicating nodes. These statistics
enhances clarity and reliability of the network. Therefore, when the correct
metrics of performance are displayed or even when a deviation is noticed, a
correction can be made


The main artefact that will be developed in
this project is a network simulation that is capable of live transmission of
video over IP. The network will be simulated to transmit different types of
video over IP such that it is possible to identify the factors that impact on
the QoS of the network for video over IP transmission. The study of the
simulated network will help the research to identify the technological and
practical bottlenecks that would arise in a real world implementation of the
network. The research will focus on the different QoS factors that are required
for unicast, multicast and broadcast transmissions.

The artefact will allow the research to
generate information that is useful from the practical point of view; it can
explore different network structures for example, and different routing
protocols, to determine the ones that are better for Video over IP transmission
and why. The network simulation allows the research to explore alternate
network designs to test different hypotheses. It also allows the researcher to
understand the way in which the network words, and further develop alternatives
that may be able to address specific problems.

4.2 Design

There are a number of different simulation
software that can be used to develop computer network simulations. NS2, OPNET
and OMNET++ are three software packages that are often used to develop computer
network simulations.

The building of implementation of the
network will begin node by node on OMNET++ platform which is in fact a C++
simulation framework that is very popular for building computer network
simulations. It can be used to simulate wired and wireless networks. It has a
huge library of extensions, which provide support for specific domains of
application such as sensor networks, Internet protocols, performance modelling,
etc., making it relatively easy to develop a network simulation using this
platform. For the current project the support available for developing IP based
networks and performance modelling would be particularly useful. It is also
open source software and therefore free to use for academic purposes. The
software also has an Integrated Development Environment, based on Eclipse, a
graphical runtime environment and other tools. OMNeT++ runs on Windows, Linux,
Mac OS X, and other Unix-like systems.




4.3 Testing Strategy

Network simulation is however processing
power intensive, and therefore the researcher would need sufficient time and
computing power to conduct the simulations. A reasonably fast personal computer
should be sufficient for conducting the simulations (Craig, 1996). The testing
strategy would be developed more fully once the literature review is completed.





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