80 research outputs found
VOIP WITH ADAPTIVE RATE IN MULTI- TRANSMISSION RATE WIRELESS LANS
âVoice over Internet Protocol (VoIP)â is a popular communication technology that plays a vital role in term of cost reduction and flexibility. However, like any emerging technology, there are still some issues with VoIP, namely providing good Quality of Service (QoS), capacity consideration and providing security. This study focuses on the QoS issue of VoIP, specifically in âWireless Local Area Networks (WLAN)â.
IEEE 802.11 is the most popular standard of wireless LANs and it offers different transmission rates for wireless channels. Different transmission rates are associated with varying available bandwidth that shall influence the transmission of VoIP traffic
VOIP WITH ADAPTIVE RATE IN MULTI- TRANSMISSION RATE WIRELESS LANS
âVoice over Internet Protocol (VoIP)â is a popular communication technology that plays a vital role in term of cost reduction and flexibility. However, like any emerging technology, there are still some issues with VoIP, namely providing good Quality of Service (QoS), capacity consideration and providing security. This study focuses on the QoS issue of VoIP, specifically in âWireless Local Area Networks (WLAN)â.
IEEE 802.11 is the most popular standard of wireless LANs and it offers different transmission rates for wireless channels. Different transmission rates are associated with varying available bandwidth that shall influence the transmission of VoIP traffic
Flow control of real-time unicast multimedia applications in best-effort networks
One of the fastest growing segments of Internet applications are real-time mul-
timedia applications, like Voice over Internet Protocol (VoIP). Real-time multimedia
applications use the User Datagram Protocol (UDP) as the transport protocol because
of the inherent conservative nature of the congestion avoidance schemes of Transmis-
sion Control Protocol (TCP). The eŸects of uncontrolled °ows on the Internet have
not yet been felt because UDP tra±c frequently constitutes only » 20% of the total
Internet tra±c. It is pertinent that real-time multimedia applications become better
citizens of the Internet, while at the same time deliver acceptable Quality of Service
(QoS).
Traditionally, packet losses and the increase in the end-to-end delay experienced
by some of the packets characterizes congestion in the network. These two signals
have been used to develop most known °ow control schemes. The current research
considers the °ow accumulation in the network as the signal for use in °ow control.
The most signiÂŻcant contribution of the current research is to propose novel end-
to-end °ow control schemes for unicast real-time multimedia °ows transmitting over
best-eÂźort networks. These control schemes are based on predictive control of the
accumulation signal. The end-to-end control schemes available in the literature are
based on reactive control that do not take into account the feedback delay existing
between the sender and the receiver nor the forward delay in the °ow dynamics. The performance of the proposed control schemes has been evaluated using the
ns-2 simulation environment. The research concludes that active control of hard real-
time °ows delivers the same or somewhat better QoS as High Bit Rate (HBR, no
control), but with a lower average bit rate. Consequently, it helps reduce bandwidth
use of controlled real-time °ows by anywhere between 31:43% to 43:96%. Proposed
reactive control schemes deliver good QoS. However, they do not scale up as well as
the predictive control schemes. Proposed predictive control schemes are eÂźective in
delivering good quality QoS while using up less bandwidth than even the reactive con-
trol schemes. They scale up well as more real-time multimedia °ows start employing
them
A novel multimedia adaptation architecture and congestion control mechanism designed for real-time interactive applications
PhDThe increasing use of interactive multimedia applications over the Internet has created a problem of congestion. This is because a majority of these applications do not respond to congestion indicators. This leads to resource starvation for responsive flows, and ultimately excessive delay and losses for all flows therefore loss of quality. This results in unfair sharing of network resources and increasing the risk of network âcongestion collapseâ.
Current Congestion Control Mechanisms such as âTCP-Friendly Rate Controlâ (TFRC) have been able to achieve âfair-shareâ of network resource when competing with responsive flows such as TCP, but TFRCâs method of congestion response (i.e. to reduce Packet Rate) is not ideally matched for interactive multimedia applications which maintain a fixed Frame Rate. This mismatch of the two rates (Packet Rate and Frame Rate) leads to buffering of frames at the Sender Buffer resulting in delay and loss, and an unacceptable reduction of quality or complete loss of service for the end-user.
To address this issue, this thesis proposes a novel Congestion Control Mechanism which is referred to as âTCP-friendly rate control â Fine Grain Scalableâ (TFGS) for interactive multimedia applications.
This new approach allows multimedia frames (data) to be sent as soon as they are generated, so that the multimedia frames can reach the destination as quickly as possible, in order to provide an isochronous interactive service. This is done by maintaining the Packet Rate of the Congestion Control Mechanism (CCM) at a level equivalent to the Frame Rate of the Multimedia Encoder.The response to congestion is to truncate the Packet Size, hence reducing the overall bitrate of the multimedia stream. This functionality of the Congestion Control Mechanism is referred to as Packet Size Truncation (PST), and takes advantage of adaptive multimedia encoding, such as Fine Grain Scalable (FGS), where the multimedia frame is encoded in order of significance, Most to Least Significant Bits. The Multimedia Adaptation Manager (MAM) truncates the multimedia frame to the size indicated by the Packet Size Truncation function of the CCM, accurately mapping user demand to available network resource. Additionally Fine Grain Scalable encoding can offer scalability at byte level granularity, providing a true match to available network resources.
This approach has the benefits of achieving a âfair-shareâ of network resource when competing with responsive flows (as similar to TFRC CCM), but it also provides an isochronous service which is of crucial benefit to real-time interactive services. Furthermore, results illustrate that an increased number of interactive multimedia flows (such as voice) can be carried over congested networks whilst maintaining a quality level equivalent to that of a standard landline telephone. This is because the loss and delay arising from the buffering of frames at the Sender Buffer is completely removed. Packets sent maintain a fixed inter-packet-gap-spacing (IPGS). This results in a majority of packets arriving at the receiving end at tight time intervals. Hence, this avoids the need of using large Playout (de-jitter) Buffer sizes and adaptive Playout Buffer configurations. As a result this reduces delay, improves interactivity and Quality of Experience (QoE) of the multimedia application
Quality of media traffic over Lossy internet protocol networks: Measurement and improvement.
Voice over Internet Protocol (VoIP) is an active area of research in the world of
communication. The high revenue made by the telecommunication companies is a
motivation to develop solutions that transmit voice over other media rather than
the traditional, circuit switching network.
However, while IP networks can carry data traffic very well due to their besteffort
nature, they are not designed to carry real-time applications such as voice.
As such several degradations can happen to the speech signal before it reaches its
destination. Therefore, it is important for legal, commercial, and technical reasons
to measure the quality of VoIP applications accurately and non-intrusively.
Several methods were proposed to measure the speech quality: some of these
methods are subjective, others are intrusive-based while others are non-intrusive.
One of the non-intrusive methods for measuring the speech quality is the E-model
standardised by the International Telecommunication Union-Telecommunication Standardisation
Sector (ITU-T).
Although the E-model is a non-intrusive method for measuring the speech quality,
but it depends on the time-consuming, expensive and hard to conduct subjective
tests to calibrate its parameters, consequently it is applicable to a limited number
of conditions and speech coders. Also, it is less accurate than the intrusive methods
such as Perceptual Evaluation of Speech Quality (PESQ) because it does not consider
the contents of the received signal.
In this thesis an approach to extend the E-model based on PESQ is proposed.
Using this method the E-model can be extended to new network conditions and
applied to new speech coders without the need for the subjective tests. The modified
E-model calibrated using PESQ is compared with the E-model calibrated using
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subjective tests to prove its effectiveness.
During the above extension the relation between quality estimation using the
E-model and PESQ is investigated and a correction formula is proposed to correct
the deviation in speech quality estimation.
Another extension to the E-model to improve its accuracy in comparison with
the PESQ looks into the content of the degraded signal and classifies packet loss
into either Voiced or Unvoiced based on the received surrounding packets. The accuracy
of the proposed method is evaluated by comparing the estimation of the new
method that takes packet class into consideration with the measurement provided
by PESQ as a more accurate, intrusive method for measuring the speech quality.
The above two extensions for quality estimation of the E-model are combined
to offer a method for estimating the quality of VoIP applications accurately, nonintrusively
without the need for the time-consuming, expensive, and hard to conduct
subjective tests.
Finally, the applicability of the E-model or the modified E-model in measuring
the quality of services in Service Oriented Computing (SOC) is illustrated
Supporting Low-Latency TCP-Based Media Streams
The dominance of the TCP protocol on the Internet and its success in maintaining Internet stability has led to several TCP-based stored media-streaming approaches. The success of these approaches raises the question whether TCP can be used for low-latency streaming. Low latency streaming allows responsive control operations for media streaming and can make interactive applications feasible. We examined adapting the TCP send buffer size based on TCP\u27s congestion window to reduce application perceived network latency. Our results show that this simple idea significantly improves the number of packets that can be delivered within 200 ms and 500 ms thresholds
Improving the Performance of Wireless LANs
This book quantifies the key factors of WLAN performance and describes methods for improvement. It provides theoretical background and empirical results for the optimum planning and deployment of indoor WLAN systems, explaining the fundamentals while supplying guidelines for design, modeling, and performance evaluation. It discusses environmental effects on WLAN systems, protocol redesign for routing and MAC, and traffic distribution; examines emerging and future network technologies; and includes radio propagation and site measurements, simulations for various network design scenarios, numerous illustrations, practical examples, and learning aids
Multimedia
The nowadays ubiquitous and effortless digital data capture and processing capabilities offered by the majority of devices, lead to an unprecedented penetration of multimedia content in our everyday life. To make the most of this phenomenon, the rapidly increasing volume and usage of digitised content requires constant re-evaluation and adaptation of multimedia methodologies, in order to meet the relentless change of requirements from both the user and system perspectives. Advances in Multimedia provides readers with an overview of the ever-growing field of multimedia by bringing together various research studies and surveys from different subfields that point out such important aspects. Some of the main topics that this book deals with include: multimedia management in peer-to-peer structures & wireless networks, security characteristics in multimedia, semantic gap bridging for multimedia content and novel multimedia applications
Satellite Communications
This study is motivated by the need to give the reader a broad view of the developments, key concepts, and technologies related to information society evolution, with a focus on the wireless communications and geoinformation technologies and their role in the environment. Giving perspective, it aims at assisting people active in the industry, the public sector, and Earth science fields as well, by providing a base for their continued work and thinking
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