The evident increase in media traffic over Internet is expected to worsen its congestion state. TCP-friendly rate control protocol TFRC is one of the most promising congestion control techniques developed so far. TFRC has been thoroughly tested in terms of being TCP-friendly, responsive, and fair. Yet, its impact on the visual quality and the peak signal-to- noise ratio PSNR of the media traffic traversing Internet is still questionable. In this paper we aimed to point out the enhancements required for TFRC that enables producing the maximum PSNR value for Internet media traffic. Firstly, we suspected the default value of n that represents the number of loss intervals used in calculating the loss event rate in the TFRC equation. This value is recommended to be set to 8 according to the latest RFC of TFRC. We investigated the effect of modifying the TFRC mechanism on the resulting PSNR of the transmitted video over Internet using TFRC via switching n across the values from 2 to 16. We investigated the effect of such variation over a simulated network environment to study its effect on the resulting PSNR for a number of arbitrary video sequences. Our simulations results showed that running TFRC with n=11 led to reaching the maximum PSNR values among all the examined values of n including its default value. Secondly, we tested the impact on the PSNR of another modification in the TFRC mechanism via switching both values of n and Nfb which is frequency of feedback messages sent by TFRC receiver to its sender every round-trip time RTT. The default value of Nfb is 1; hence we scanned every possible combination of n and Nfb ranging from 2 to 16, and from 1 to 4, respectively and recorded the produced PSNR. It was obvious that several other combinations of n and Nfb produced higher PSNR values other than their default values in the request for comment RFC of TFRC. We hereby suggest using an enhanced TFRC that we abbreviated as ETFRC which has the values of n and Nfb value set to 4 and 11 respectively as a replacement for the traditional TFRC to enable reaching higher PSNR for media traffic over Internet.
Multicast protocols can be classified as either reliable or unreliable both uses a best effort method of
setting up, maintaining and tearing down the multicast distribution tree. Pragmatic General Multicast (PGM),
Reliable Hybrid Multicast Protocol (RHMP) and Elastic Reliable Multicast (ERM) are examples of reliable
multicast protocols. PGM and ERM sends flood messages to the Rendezvous Point source (RPS) from the
source node towards the stub nodes which then forward it to leaf nodes, leaf nodes that are not interested sends
a prune message while any leaf node that misses a packet sends a message to the RPS through the stub node
requesting for the missed multicast packet. A repair multicast packet is then forwarded to all leaf nodes that
requested for it. For a large distribution tree congestion might occur in the RPS if there is much failure and the
leaf node keeps requesting for repair data. In a distributed RPS such as the reliable hybrid multicast protocol
(RHMP) the stub nodes originates the flood message to the leaf and uninterested leaf sends prune message, any
stub that has one or more interested leaf sends a join message to the RPS. If a leaf node in the multicast
distribution misses a multicast packet it requests a repair packet its stub node sends the repair data. A
simulation model was developed to mimic the behaviour of PGM, ERM and RHMP in different network size
using hierarchical network and the control bandwidth overhead (CBO) for each of the multicast protocols was
calculated at the source node, stud node and leaf nodes, CBO was use as the cost metric. The result shows that
the RHMP uses less CBO than PGM and ERM in a sparsely and densely populated network at the source and
leaf nodes but more CBO is used at the stub nodes but since more than one stub nodes act as the RPS to the
leafs connected to it, RHMP was found out to be better than the PGM or ERM.
Recital Study of Various Congestion Control Protocols in wireless networkiosrjce
IOSR Journal of Computer Engineering (IOSR-JCE) is a double blind peer reviewed International Journal that provides rapid publication (within a month) of articles in all areas of computer engineering and its applications. The journal welcomes publications of high quality papers on theoretical developments and practical applications in computer technology. Original research papers, state-of-the-art reviews, and high quality technical notes are invited for publications.
PERFORMANCE EVALUATION OF SELECTED E2E TCP CONGESTION CONTROL MECHANISM OVER ...ijwmn
TCP is one of the main protocols that govern the Internet traffic nowadays. However, it suffers significant
performance degradation over wireless links. Since wireless networks are leading the communication
technologies recently, it is imperative to introduce effective solutions for the TCP congestion control
mechanisms over such networks. In this research four End-to-End TCP implementations are discussed,
they are TCP Westwood, Hybla, Highspeed, and NewReno. The performance of these variants is compared
using LTE emulated environment in terms of throughput, delay, and fairness. Ns-3 simulator is used to
simulate the LTE networks environment. The simulation results showed that TCP Highspeed achieves the
best throughput results. Although TCP Westwood recorded the lowest latency values comparing to others,
it behaved unfairly among different traffic flows. Moreover, TCP Hybla demonstrated the best fairness
behaviour among other TCP variants
Performance comparison of two clipping based filtering methods for papr reduc...ijmnct
The growth of wireless communication technologies has been producing the intense demand for high-speed,
efficient, reliable voice & data communication. As a result, third generation partnership project (3GPP)
has implemented next generation wireless communication technology long term evolution (LTE) which is
designed to increase the capacity and speed of existing mobile telephone & data networks. LTE has
adopted a multicarrier transmission technique known as orthogonal frequency division multiplexing
(OFDM). OFDM meets the LTE requirement for spectrum flexibility and enables cost-efficient solutions for
very wide carriers. One major generic problem of OFDM technique is high peak to average power ratio
(PAPR) which is defined as the ratio of the peak power to the average power of the OFDM signal. A
trade-off is necessary for reducing PAPR with increasing bit error rate (BER), computational complexity or
data rate loss etc. In this paper, two clipping based filtering methods have been implemented & also
analyzed their modulation effects on reducing PAPR.
PERFORMANCE ANALYSIS OF MULTI-PATH TCP NETWORKIJCNCJournal
MPTCP is proposed by IETF working group, it allows a single TCP stream to be split across multiple
paths. It has obvious benefits in performance and reliability. MPTCP has implemented in Linux-based
distributions that can be compiled and installed to be used for both real and experimental scenarios. In this
article, we provide performance analyses for MPTCP with a laptop connected to WiFi access point and 3G
cellular network at the same time. We prove experimentally that MPTCP outperforms regular TCP for
WiFi or 3G interfaces. We also compare four types of congestion control algorithms for MPTCP that are
also implemented in the Linux Kernel. Results show that Alias Linked Increase Congestion Control
algorithm outperforms the others in the normal traffic load while Balanced Linked Adaptation algorithm
outperforms the rest when the paths are shared with heavy traffic, which is not supported by MPTCP.
To accommodate real-time multimedia applications that achieve Quality of Service (QOS) requirements in
a wireless ad-hoc network, a QOS control mechanisms is needed. Correspondence over such networks
must consider other aspects in regard to network properties; that the time it takes to send a message and
reach its destination faces different variables from one message to the other in a short time. Therefore, the
equation of calculating the time required to resend the message must be able to contain the worst case and
acknowledges different features for the network. The objective of this paper is to improve retransmission
time calculation adaptability when occurring using Transmission Control Protocol (TCP) over wireless Ad
hoc networks. Hence, it aims to obtain more accurate time to ensure retransmission time occurring in
accordance to the network environment variables efficiently.
Comparative performance analysis of different modulation techniques for papr ...IJCNCJournal
One of the most important multi-carrier tran
smission techniques used in the latest wireless com
munication
arena is known as Orthogonal Frequency Division Mul
tiplexing (OFDM). It has several characteristics
such as providing greater immunity to multipath fad
ing & impulse noise, eliminating Inter Symbol
Interference (ISI) & Inter Carrier Interference (IC
I) using a guard interval known as Cyclic Prefix (C
P). A
regular difficulty of OFDM signal is high peak to a
verage power ratio (PAPR) which is defined as the r
atio
of the peak power to the average power of OFDM Sign
al. An improved design of amplitude clipping &
filtering technique of us previously reduced signif
icant amount of PAPR with slightly increase bit err
or rate
(BER) compare to an existing method in case of Quad
rature Phase Shift Keying (QPSK) & Quadrature
Amplitude Modulation (QAM). This paper investigates
a comparative performance analysis of the differen
t
higher order modulation techniques on that design.
Multicast protocols can be classified as either reliable or unreliable both uses a best effort
method of setting up, maintaining and tearing down the multicast distribution tree. Pragmatic
General Multicast (PGM), Reliable Hybrid Multicast Protocol (RHMP) and Elastic Reliable
Multicast (ERM) are examples of reliable multicast protocols. PGM and ERM sends flood messages
to the Rendezvous Point source (RPS) from the source node towards the stub nodes which then
forward it to leaf nodes, leaf nodes that are not interested sends a prune message while any leaf node
that misses a packet sends a message to the RPS through the stub node requesting for the missed
multicast packet. A repair multicast packet is then forwarded to all leaf nodes that requested for it.
For a large distribution tree congestion might occur in the RPS if there is much failure and the leaf
node keeps requesting for repair data. In a distributed RPS such as the reliable hybrid multicast
protocol (RHMP) the stub nodes originates the flood message to the leaf and uninterested leaf sends
prune message, any stub that has one or more interested leaf sends a join message to the RPS. If a
leaf node in the multicast distribution misses a multicast packet it requests a repair packet its stub
node sends the repair data. A simulation model was developed to mimic the behaviour of PGM, ERM
and RHMP in different network size using hierarchical network and the control bandwidth overhead
(CBO) for each of the multicast protocols was calculated at the source node, stud node and leaf
nodes, CBO was use as the cost metric. The result shows that the RHMP uses less CBO than PGM
and ERM in a sparsely and densely populated network at the source and leaf nodes but more CBO is
used at the stub nodes but since more than one stub nodes act as the RPS to the leafs connected to it,
RHMP was found out to be better than the PGM or ERM.
Multicast protocols can be classified as either reliable or unreliable both uses a best effort method of
setting up, maintaining and tearing down the multicast distribution tree. Pragmatic General Multicast (PGM),
Reliable Hybrid Multicast Protocol (RHMP) and Elastic Reliable Multicast (ERM) are examples of reliable
multicast protocols. PGM and ERM sends flood messages to the Rendezvous Point source (RPS) from the
source node towards the stub nodes which then forward it to leaf nodes, leaf nodes that are not interested sends
a prune message while any leaf node that misses a packet sends a message to the RPS through the stub node
requesting for the missed multicast packet. A repair multicast packet is then forwarded to all leaf nodes that
requested for it. For a large distribution tree congestion might occur in the RPS if there is much failure and the
leaf node keeps requesting for repair data. In a distributed RPS such as the reliable hybrid multicast protocol
(RHMP) the stub nodes originates the flood message to the leaf and uninterested leaf sends prune message, any
stub that has one or more interested leaf sends a join message to the RPS. If a leaf node in the multicast
distribution misses a multicast packet it requests a repair packet its stub node sends the repair data. A
simulation model was developed to mimic the behaviour of PGM, ERM and RHMP in different network size
using hierarchical network and the control bandwidth overhead (CBO) for each of the multicast protocols was
calculated at the source node, stud node and leaf nodes, CBO was use as the cost metric. The result shows that
the RHMP uses less CBO than PGM and ERM in a sparsely and densely populated network at the source and
leaf nodes but more CBO is used at the stub nodes but since more than one stub nodes act as the RPS to the
leafs connected to it, RHMP was found out to be better than the PGM or ERM.
Recital Study of Various Congestion Control Protocols in wireless networkiosrjce
IOSR Journal of Computer Engineering (IOSR-JCE) is a double blind peer reviewed International Journal that provides rapid publication (within a month) of articles in all areas of computer engineering and its applications. The journal welcomes publications of high quality papers on theoretical developments and practical applications in computer technology. Original research papers, state-of-the-art reviews, and high quality technical notes are invited for publications.
PERFORMANCE EVALUATION OF SELECTED E2E TCP CONGESTION CONTROL MECHANISM OVER ...ijwmn
TCP is one of the main protocols that govern the Internet traffic nowadays. However, it suffers significant
performance degradation over wireless links. Since wireless networks are leading the communication
technologies recently, it is imperative to introduce effective solutions for the TCP congestion control
mechanisms over such networks. In this research four End-to-End TCP implementations are discussed,
they are TCP Westwood, Hybla, Highspeed, and NewReno. The performance of these variants is compared
using LTE emulated environment in terms of throughput, delay, and fairness. Ns-3 simulator is used to
simulate the LTE networks environment. The simulation results showed that TCP Highspeed achieves the
best throughput results. Although TCP Westwood recorded the lowest latency values comparing to others,
it behaved unfairly among different traffic flows. Moreover, TCP Hybla demonstrated the best fairness
behaviour among other TCP variants
Performance comparison of two clipping based filtering methods for papr reduc...ijmnct
The growth of wireless communication technologies has been producing the intense demand for high-speed,
efficient, reliable voice & data communication. As a result, third generation partnership project (3GPP)
has implemented next generation wireless communication technology long term evolution (LTE) which is
designed to increase the capacity and speed of existing mobile telephone & data networks. LTE has
adopted a multicarrier transmission technique known as orthogonal frequency division multiplexing
(OFDM). OFDM meets the LTE requirement for spectrum flexibility and enables cost-efficient solutions for
very wide carriers. One major generic problem of OFDM technique is high peak to average power ratio
(PAPR) which is defined as the ratio of the peak power to the average power of the OFDM signal. A
trade-off is necessary for reducing PAPR with increasing bit error rate (BER), computational complexity or
data rate loss etc. In this paper, two clipping based filtering methods have been implemented & also
analyzed their modulation effects on reducing PAPR.
PERFORMANCE ANALYSIS OF MULTI-PATH TCP NETWORKIJCNCJournal
MPTCP is proposed by IETF working group, it allows a single TCP stream to be split across multiple
paths. It has obvious benefits in performance and reliability. MPTCP has implemented in Linux-based
distributions that can be compiled and installed to be used for both real and experimental scenarios. In this
article, we provide performance analyses for MPTCP with a laptop connected to WiFi access point and 3G
cellular network at the same time. We prove experimentally that MPTCP outperforms regular TCP for
WiFi or 3G interfaces. We also compare four types of congestion control algorithms for MPTCP that are
also implemented in the Linux Kernel. Results show that Alias Linked Increase Congestion Control
algorithm outperforms the others in the normal traffic load while Balanced Linked Adaptation algorithm
outperforms the rest when the paths are shared with heavy traffic, which is not supported by MPTCP.
To accommodate real-time multimedia applications that achieve Quality of Service (QOS) requirements in
a wireless ad-hoc network, a QOS control mechanisms is needed. Correspondence over such networks
must consider other aspects in regard to network properties; that the time it takes to send a message and
reach its destination faces different variables from one message to the other in a short time. Therefore, the
equation of calculating the time required to resend the message must be able to contain the worst case and
acknowledges different features for the network. The objective of this paper is to improve retransmission
time calculation adaptability when occurring using Transmission Control Protocol (TCP) over wireless Ad
hoc networks. Hence, it aims to obtain more accurate time to ensure retransmission time occurring in
accordance to the network environment variables efficiently.
Comparative performance analysis of different modulation techniques for papr ...IJCNCJournal
One of the most important multi-carrier tran
smission techniques used in the latest wireless com
munication
arena is known as Orthogonal Frequency Division Mul
tiplexing (OFDM). It has several characteristics
such as providing greater immunity to multipath fad
ing & impulse noise, eliminating Inter Symbol
Interference (ISI) & Inter Carrier Interference (IC
I) using a guard interval known as Cyclic Prefix (C
P). A
regular difficulty of OFDM signal is high peak to a
verage power ratio (PAPR) which is defined as the r
atio
of the peak power to the average power of OFDM Sign
al. An improved design of amplitude clipping &
filtering technique of us previously reduced signif
icant amount of PAPR with slightly increase bit err
or rate
(BER) compare to an existing method in case of Quad
rature Phase Shift Keying (QPSK) & Quadrature
Amplitude Modulation (QAM). This paper investigates
a comparative performance analysis of the differen
t
higher order modulation techniques on that design.
Multicast protocols can be classified as either reliable or unreliable both uses a best effort
method of setting up, maintaining and tearing down the multicast distribution tree. Pragmatic
General Multicast (PGM), Reliable Hybrid Multicast Protocol (RHMP) and Elastic Reliable
Multicast (ERM) are examples of reliable multicast protocols. PGM and ERM sends flood messages
to the Rendezvous Point source (RPS) from the source node towards the stub nodes which then
forward it to leaf nodes, leaf nodes that are not interested sends a prune message while any leaf node
that misses a packet sends a message to the RPS through the stub node requesting for the missed
multicast packet. A repair multicast packet is then forwarded to all leaf nodes that requested for it.
For a large distribution tree congestion might occur in the RPS if there is much failure and the leaf
node keeps requesting for repair data. In a distributed RPS such as the reliable hybrid multicast
protocol (RHMP) the stub nodes originates the flood message to the leaf and uninterested leaf sends
prune message, any stub that has one or more interested leaf sends a join message to the RPS. If a
leaf node in the multicast distribution misses a multicast packet it requests a repair packet its stub
node sends the repair data. A simulation model was developed to mimic the behaviour of PGM, ERM
and RHMP in different network size using hierarchical network and the control bandwidth overhead
(CBO) for each of the multicast protocols was calculated at the source node, stud node and leaf
nodes, CBO was use as the cost metric. The result shows that the RHMP uses less CBO than PGM
and ERM in a sparsely and densely populated network at the source and leaf nodes but more CBO is
used at the stub nodes but since more than one stub nodes act as the RPS to the leafs connected to it,
RHMP was found out to be better than the PGM or ERM.
International Journal of Engineering Research and Applications (IJERA) is a team of researchers not publication services or private publications running the journals for monetary benefits, we are association of scientists and academia who focus only on supporting authors who want to publish their work. The articles published in our journal can be accessed online, all the articles will be archived for real time access.
Our journal system primarily aims to bring out the research talent and the works done by sciaentists, academia, engineers, practitioners, scholars, post graduate students of engineering and science. This journal aims to cover the scientific research in a broader sense and not publishing a niche area of research facilitating researchers from various verticals to publish their papers. It is also aimed to provide a platform for the researchers to publish in a shorter of time, enabling them to continue further All articles published are freely available to scientific researchers in the Government agencies,educators and the general public. We are taking serious efforts to promote our journal across the globe in various ways, we are sure that our journal will act as a scientific platform for all researchers to publish their works online.
Fault tolerant wireless sensor mac protocol for efficient collision avoidancegraphhoc
In sensor networks communication by broadcast methods involves many hazards, especially collision. Several MAC layer protocols have been proposed to resolve the problem of collision namely ARBP, where the best achieved success rate is 90%. We hereby propose a MAC protocol which achieves a greater success rate (Success rate is defined as the percentage of delivered packets at the source reaching the destination successfully) by reducing the number of collisions, but by trading off the average propagation delay of transmission. Our proposed protocols are also shown to be more energy efficient in terms of energy dissipation per message delivery, compared to the currently existing protocol.
Long-Term Evolution (LTE), an emerging and promising fourth generation mobile technology, is expected
to offer ubiquitous broadband access to the mobile subscribers. In this paper, the performance of Frame
Level Scheduler (FLS), Exponential (EXP) rule, Logarithmic (LOG) rule and Maximum-Largest Weighted
Delay First (M-LWDF) packet scheduling algorithms has been studied in the downlink 3GPP LTE cellular
network. To this aim, a single cell with interference scenario has been considered. The performance
evaluation is made by varying the number of UEs ranging from 10 to 50 (Case 1) and user speed in the
range of [3, 120] km/h (Case 2). Results show that while the number of UEs and user speed increases, the
performance of the considered scheduling schemes degrades and in both case FLS outperforms other three
schemes in terms of several performance indexes such as average throughput, packet loss ratio (PLR),
packet delay and fairness index.
Macro with pico cells (hetnets) system behaviour using well known scheduling ...ijwmn
This paper demonstrates the concept of using Heterogeneous networks (HetNets) to improve Long Term Evolution (LTE) system by introducing the LTE Advance (LTE-A). The type of HetNets that has been chosen for this study is Macro with Pico cells. Comparing the system performance with and without Pico cells has clearly illustrated using three well-known scheduling algorithms (Proportional Fair PF, Maximum Largest Weighted Delay First MLWDF and Exponential/Proportional Fair EXP/PF). The system is judged based on throughput, Packet Loss Ratio PLR, delay and fairness.. A simulation platform called LTE-Sim has been used to collect the data and produce the paper’s outcomes and graphs. The results prove that adding Pico cells enhances the overall system performance. From the simulation outcomes, the overall system performance is as follows: throughput is duplicated or tripled based on the number of users, the PLR is almost quartered, the delay is nearly reduced ten times (PF case) and changed to be a half (MLWDF/EXP cases), and the fairness stays closer to value of 1. It is considered an efficient and cost effective way to increase the throughput, coverage and reduce the latency.
"Performance Evaluation and Comparison of Westwood+, New Reno and Vegas TCP ...losalamos
Luigi A. Grieco, Saverio Mascolo.
ACM CCR, Vol.34 No.2, April 2004.
This article aims at evaluating a comparison between three TCP congestion control algorithms. A really interesting reading.
A Review Paper on: The PAPR Analysis of MIMO-OFDM SystemsIJEEE
Orthogonal frequency division Multiplexing (OFDM) is one of the most popular multicarrier or multiplexing modulation techniques which transmits many signals over a single path in high speed wireless communication. OFDM convert high data rate stream in to smaller data rate stream. Due to this high data rate and ability to combat frequency selective fading, OFDM has a strong candidate for 4G wireless networks. Because of OFDM-MIMO advantages in multipath fading channel e.g. stout against ISI, ICI and some other advantages like best QoS for multiple users, efficient convention of bandwidth it is suggested to be the modulation technique for next generation 4G networks e.g. LTE. OFDM combined with multiple-input multiple-output (MIMO) to increase system capacity over the time variant frequency-selective channels and the diversity gain. The radio transmitter stations for covering and getting enough transmitted power in their desired area has to use High Power Amplifier (HPA) operable near to the saturation region or else, an aspect memory-less nonlinear distortion will affect the communication path. But along with all its advantages there are some disadvantages also. e.g. High PAPR (Peak to Average Power Ratio) at the transmitter end and BER (Bit Error Rate) at the receiving end. As OFDM is only used in the downlink of 4G networks. To reduce the problems of OFDM-MIMO systems some procedures as SLM, PTS, Clipping, Coding, & Pre-coding etc are suggested but none of them is proficient enough to reduce the PAPR and BER up to standard value. This Paper will discuss some techniques of PAPR reduction, and both their advantages and disadvantages.
avoiding retransmissions using random coding scheme or fountain code schemeIJAEMSJORNAL
In a perfect world, the throughput of a Multipath TCP (MPTCP) association ought to be as high as that of different disjoint single-way TCP streams. In actuality, the throughput of MPTCP is far lower than anticipated. In this paper, we lead angeneral reproduction construct ponder in light of this peculiarity, and the outcomes show that a sub stream encountering high postponement and misfortune extremely influences the execution of other sub streams, in this manner turning into the half back of the MPTCP association and knowingly humiliating the total great put. To handle this issue, we propose Wellspring code-based Multipath TCP (FMTCP), which viably mitigates the negative effect of the heterogeneity of modified ways. FMTCP exploits the unintentional way of the wellspring code to adaptably transmit encoded images from the same or diverse information hinders over various sub streams. In addition, we plan an information portion calculation in view of the foreseen bundle arriving time and deciphering command to facilitate the transmissions of various sub streams. Quantitative reviews are given to demonstrate the advantage of FMTCP. We likewise assess the presentation of FMTCP through ns-2 recreations and exhibit that FMTCP beats IETF-MPTCP, a run of the mill MPTCP approach, when the ways have various misfortune and deferral as far as higher aggregate great put, bring down postponement, and jitter. Also, FMTCP acknowledges high security under sudden changes of way quality.
International Journal of Engineering Research and Applications (IJERA) is an open access online peer reviewed international journal that publishes research and review articles in the fields of Computer Science, Neural Networks, Electrical Engineering, Software Engineering, Information Technology, Mechanical Engineering, Chemical Engineering, Plastic Engineering, Food Technology, Textile Engineering, Nano Technology & science, Power Electronics, Electronics & Communication Engineering, Computational mathematics, Image processing, Civil Engineering, Structural Engineering, Environmental Engineering, VLSI Testing & Low Power VLSI Design etc.
The four basic Radio Resource Management (RRM) measurements in Long Term Evolution (LTE) system
are Channel Quality Indicator (CQI), Reference Signal Received Power (RSRP), Reference Signal
Received Quality (RSRQ), and Carrier Received Signal Strength Indicator (RSSI). A measurement of channel quality represented by Signal to Interference plus Noise Ratio (SINR) is used for link adaptation along with packet scheduling, whereas RSRP and RSRQ are needed for making handover decision during intra-eUTRAN (evolved Universal Terrestrial Random Access Network) handover in LTE. In this paper,
some practical measurement results recorded from a live LTE network of Australia using a commercial measurement tool namely NEMO Handy are analysed to verify the possible relationships among SINR, RSRP, RSSI and RSRQ as well as to evaluate the effects of SNR on throughput. In addition, the intraeUTRAN handover events occurred during the test period within the test area are studied. The analysis
yields some useful information such as: if the SINR is good for a measurement slot, higher throughput is
achieved; RSRP and SNR are proportional to each other on average; and lesser is the difference between
RSSI and RSRP, better is the RSRQ – each of which is consistent with theory. All the measurement results
are evaluated using computer programs built on MATLAB platform.
OPTIMUM EFFICIENT MOBILITY MANAGEMENT SCHEME FOR IPv6 ijngnjournal
Mobile IPv6 (MIPv6) and Hierarchical Mobile IPv6 (HMIPv6) both are the mobility management solutions proposed by the Internet Engineering Task Force (IETF) to support IP Mobility. It’s been an important issue, that upon certain condition, out of MIPv6 and HMIPv6 which one is better. In this paper an Optimum Efficient Mobility Management (OEMM) scheme is described on the basis of analytical model which shows that OEMM Scheme is better in terms of performance and applicability of MIPv6 and HMIPv6. It shows that which one is better alternative between MIPv6 and HMIPv6 and if HMIPv6 is adopted it chooses the best Mobility Anchor Point (MAP). Finally it is illustrated that OEMM scheme is
better than that of MIPv6 and HMIPv6.
Available network bandwidth schema to improve performance in tcp protocolsIJCNCJournal
The TCP congestion control mechanism in standard implementations presents several problems, for
example, large queue lengths in network routers and packet losses, a misleading reduce of the transmission
rate when there are link failures, among others. This paper proposes a schema to congestion control in
TCP protocols, called NGWA, witch is based on the network bandwidth. The NGWA provides information
considering the available bandwidth of the network infrastructure to the endpoints of the TCP connection.
Hence, it helps in choosing a better transmission rate for TCP improving its performance. Simulation
results show superior performance of the proposed scheme when compared to those obtained by TCP New
Reno and standard TCP. A physical implementation in the Linux kernel was performed to prove the correct
operation of the proposal.
Analisa AC atau sering disebut analisa sinyal kecil pada penguat adalah analisa penguat sinyal AC, dengan memblok sinyal DC yaitu dengan memberikan kapasitor coupling pada sinyal input dan sinyal output.Pada analisa AC untuk frekuensi midband/passband.
International Journal of Engineering Research and Applications (IJERA) is a team of researchers not publication services or private publications running the journals for monetary benefits, we are association of scientists and academia who focus only on supporting authors who want to publish their work. The articles published in our journal can be accessed online, all the articles will be archived for real time access.
Our journal system primarily aims to bring out the research talent and the works done by sciaentists, academia, engineers, practitioners, scholars, post graduate students of engineering and science. This journal aims to cover the scientific research in a broader sense and not publishing a niche area of research facilitating researchers from various verticals to publish their papers. It is also aimed to provide a platform for the researchers to publish in a shorter of time, enabling them to continue further All articles published are freely available to scientific researchers in the Government agencies,educators and the general public. We are taking serious efforts to promote our journal across the globe in various ways, we are sure that our journal will act as a scientific platform for all researchers to publish their works online.
Fault tolerant wireless sensor mac protocol for efficient collision avoidancegraphhoc
In sensor networks communication by broadcast methods involves many hazards, especially collision. Several MAC layer protocols have been proposed to resolve the problem of collision namely ARBP, where the best achieved success rate is 90%. We hereby propose a MAC protocol which achieves a greater success rate (Success rate is defined as the percentage of delivered packets at the source reaching the destination successfully) by reducing the number of collisions, but by trading off the average propagation delay of transmission. Our proposed protocols are also shown to be more energy efficient in terms of energy dissipation per message delivery, compared to the currently existing protocol.
Long-Term Evolution (LTE), an emerging and promising fourth generation mobile technology, is expected
to offer ubiquitous broadband access to the mobile subscribers. In this paper, the performance of Frame
Level Scheduler (FLS), Exponential (EXP) rule, Logarithmic (LOG) rule and Maximum-Largest Weighted
Delay First (M-LWDF) packet scheduling algorithms has been studied in the downlink 3GPP LTE cellular
network. To this aim, a single cell with interference scenario has been considered. The performance
evaluation is made by varying the number of UEs ranging from 10 to 50 (Case 1) and user speed in the
range of [3, 120] km/h (Case 2). Results show that while the number of UEs and user speed increases, the
performance of the considered scheduling schemes degrades and in both case FLS outperforms other three
schemes in terms of several performance indexes such as average throughput, packet loss ratio (PLR),
packet delay and fairness index.
Macro with pico cells (hetnets) system behaviour using well known scheduling ...ijwmn
This paper demonstrates the concept of using Heterogeneous networks (HetNets) to improve Long Term Evolution (LTE) system by introducing the LTE Advance (LTE-A). The type of HetNets that has been chosen for this study is Macro with Pico cells. Comparing the system performance with and without Pico cells has clearly illustrated using three well-known scheduling algorithms (Proportional Fair PF, Maximum Largest Weighted Delay First MLWDF and Exponential/Proportional Fair EXP/PF). The system is judged based on throughput, Packet Loss Ratio PLR, delay and fairness.. A simulation platform called LTE-Sim has been used to collect the data and produce the paper’s outcomes and graphs. The results prove that adding Pico cells enhances the overall system performance. From the simulation outcomes, the overall system performance is as follows: throughput is duplicated or tripled based on the number of users, the PLR is almost quartered, the delay is nearly reduced ten times (PF case) and changed to be a half (MLWDF/EXP cases), and the fairness stays closer to value of 1. It is considered an efficient and cost effective way to increase the throughput, coverage and reduce the latency.
"Performance Evaluation and Comparison of Westwood+, New Reno and Vegas TCP ...losalamos
Luigi A. Grieco, Saverio Mascolo.
ACM CCR, Vol.34 No.2, April 2004.
This article aims at evaluating a comparison between three TCP congestion control algorithms. A really interesting reading.
A Review Paper on: The PAPR Analysis of MIMO-OFDM SystemsIJEEE
Orthogonal frequency division Multiplexing (OFDM) is one of the most popular multicarrier or multiplexing modulation techniques which transmits many signals over a single path in high speed wireless communication. OFDM convert high data rate stream in to smaller data rate stream. Due to this high data rate and ability to combat frequency selective fading, OFDM has a strong candidate for 4G wireless networks. Because of OFDM-MIMO advantages in multipath fading channel e.g. stout against ISI, ICI and some other advantages like best QoS for multiple users, efficient convention of bandwidth it is suggested to be the modulation technique for next generation 4G networks e.g. LTE. OFDM combined with multiple-input multiple-output (MIMO) to increase system capacity over the time variant frequency-selective channels and the diversity gain. The radio transmitter stations for covering and getting enough transmitted power in their desired area has to use High Power Amplifier (HPA) operable near to the saturation region or else, an aspect memory-less nonlinear distortion will affect the communication path. But along with all its advantages there are some disadvantages also. e.g. High PAPR (Peak to Average Power Ratio) at the transmitter end and BER (Bit Error Rate) at the receiving end. As OFDM is only used in the downlink of 4G networks. To reduce the problems of OFDM-MIMO systems some procedures as SLM, PTS, Clipping, Coding, & Pre-coding etc are suggested but none of them is proficient enough to reduce the PAPR and BER up to standard value. This Paper will discuss some techniques of PAPR reduction, and both their advantages and disadvantages.
avoiding retransmissions using random coding scheme or fountain code schemeIJAEMSJORNAL
In a perfect world, the throughput of a Multipath TCP (MPTCP) association ought to be as high as that of different disjoint single-way TCP streams. In actuality, the throughput of MPTCP is far lower than anticipated. In this paper, we lead angeneral reproduction construct ponder in light of this peculiarity, and the outcomes show that a sub stream encountering high postponement and misfortune extremely influences the execution of other sub streams, in this manner turning into the half back of the MPTCP association and knowingly humiliating the total great put. To handle this issue, we propose Wellspring code-based Multipath TCP (FMTCP), which viably mitigates the negative effect of the heterogeneity of modified ways. FMTCP exploits the unintentional way of the wellspring code to adaptably transmit encoded images from the same or diverse information hinders over various sub streams. In addition, we plan an information portion calculation in view of the foreseen bundle arriving time and deciphering command to facilitate the transmissions of various sub streams. Quantitative reviews are given to demonstrate the advantage of FMTCP. We likewise assess the presentation of FMTCP through ns-2 recreations and exhibit that FMTCP beats IETF-MPTCP, a run of the mill MPTCP approach, when the ways have various misfortune and deferral as far as higher aggregate great put, bring down postponement, and jitter. Also, FMTCP acknowledges high security under sudden changes of way quality.
International Journal of Engineering Research and Applications (IJERA) is an open access online peer reviewed international journal that publishes research and review articles in the fields of Computer Science, Neural Networks, Electrical Engineering, Software Engineering, Information Technology, Mechanical Engineering, Chemical Engineering, Plastic Engineering, Food Technology, Textile Engineering, Nano Technology & science, Power Electronics, Electronics & Communication Engineering, Computational mathematics, Image processing, Civil Engineering, Structural Engineering, Environmental Engineering, VLSI Testing & Low Power VLSI Design etc.
The four basic Radio Resource Management (RRM) measurements in Long Term Evolution (LTE) system
are Channel Quality Indicator (CQI), Reference Signal Received Power (RSRP), Reference Signal
Received Quality (RSRQ), and Carrier Received Signal Strength Indicator (RSSI). A measurement of channel quality represented by Signal to Interference plus Noise Ratio (SINR) is used for link adaptation along with packet scheduling, whereas RSRP and RSRQ are needed for making handover decision during intra-eUTRAN (evolved Universal Terrestrial Random Access Network) handover in LTE. In this paper,
some practical measurement results recorded from a live LTE network of Australia using a commercial measurement tool namely NEMO Handy are analysed to verify the possible relationships among SINR, RSRP, RSSI and RSRQ as well as to evaluate the effects of SNR on throughput. In addition, the intraeUTRAN handover events occurred during the test period within the test area are studied. The analysis
yields some useful information such as: if the SINR is good for a measurement slot, higher throughput is
achieved; RSRP and SNR are proportional to each other on average; and lesser is the difference between
RSSI and RSRP, better is the RSRQ – each of which is consistent with theory. All the measurement results
are evaluated using computer programs built on MATLAB platform.
OPTIMUM EFFICIENT MOBILITY MANAGEMENT SCHEME FOR IPv6 ijngnjournal
Mobile IPv6 (MIPv6) and Hierarchical Mobile IPv6 (HMIPv6) both are the mobility management solutions proposed by the Internet Engineering Task Force (IETF) to support IP Mobility. It’s been an important issue, that upon certain condition, out of MIPv6 and HMIPv6 which one is better. In this paper an Optimum Efficient Mobility Management (OEMM) scheme is described on the basis of analytical model which shows that OEMM Scheme is better in terms of performance and applicability of MIPv6 and HMIPv6. It shows that which one is better alternative between MIPv6 and HMIPv6 and if HMIPv6 is adopted it chooses the best Mobility Anchor Point (MAP). Finally it is illustrated that OEMM scheme is
better than that of MIPv6 and HMIPv6.
Available network bandwidth schema to improve performance in tcp protocolsIJCNCJournal
The TCP congestion control mechanism in standard implementations presents several problems, for
example, large queue lengths in network routers and packet losses, a misleading reduce of the transmission
rate when there are link failures, among others. This paper proposes a schema to congestion control in
TCP protocols, called NGWA, witch is based on the network bandwidth. The NGWA provides information
considering the available bandwidth of the network infrastructure to the endpoints of the TCP connection.
Hence, it helps in choosing a better transmission rate for TCP improving its performance. Simulation
results show superior performance of the proposed scheme when compared to those obtained by TCP New
Reno and standard TCP. A physical implementation in the Linux kernel was performed to prove the correct
operation of the proposal.
Analisa AC atau sering disebut analisa sinyal kecil pada penguat adalah analisa penguat sinyal AC, dengan memblok sinyal DC yaitu dengan memberikan kapasitor coupling pada sinyal input dan sinyal output.Pada analisa AC untuk frekuensi midband/passband.
ВКонтакте – это самая крупная социальная сеть в Казахстане – более 6 млн активных пользователей в месяц!
Admixer – официальный селлер рекламных возможностей в Казахстане.
Marketing to Asia - What to Know, What to DoJosh Steimle
This presentation accompanies an article I wrote for Entrepreneur Magazine at http://www.entrepreneur.com/article/252992. I've been running a digital marketing agency in Hong Kong for the past 3 years and have done business in China, Japan, Singapore, Korea, Vietnam, Singapore, and more. Here are some of the things I've learned about doing business in Asia, and some tips when it comes to marketing in the region.
In the last few years, video streaming facilities over TCP or UDP, such as YouTube, Facetime, Daily-motion, Mobile video calling have become more and more popular. The important
challenge in streaming broadcasting over the Internet is to spread the uppermost potential quality,
observe to the broadcasting play out time limitation, and efficiently and equally share the offered
bandwidth with TCP or UDP, and additional traffic types. This work familiarizes the Streaming
Media Data Congestion Control protocol (SMDCC), a new adaptive broadcasting streaming
congestion management protocol in which the connection’s data packets transmission frequency is
adjusted allowing to the dynamic bandwidth share of connection using SMDCC, the bandwidth share
of a connection is projected using algorithms similar to those introduced in TCP Westwood. SMDCC
avoids the Slow Jump phase in TCP. As a result, SMDCC does not show the pronounced rate
alternations distinguishing of modern TCP, so providing congestion control that is more appropriate
for streaming broadcasting applications. Besides, SMDCC is fair, sharing the bandwidth equitably
among a set of SMDCC connections. Main benefit is robustness when packet harms are due to
indiscriminate errors, which is typical of wireless links and is becoming an increasing concern due to
the emergence of wireless Internet access. In the presence of indiscriminate errors, SMDCC is also
approachable to TCP Tahoe and Reno (TTR). We provide simulation results using the ns3 simulator
for our protocol running together with TCP Tahoe and Reno.
VEGAS: Better Performance than other TCP Congestion Control Algorithms on MANETsCSCJournals
The wireless communication TCP/IP protocol is an important role in developing communication systems and which provides better and reliable communication capabilities in almost all kinds of networking environment. The wireless networking technology and the new kind of requirements in communication systems need some extensions to the original design of TCP for on coming technology development. In this paper we have analyzed six TCP Congestion Control Algorithms and their performance on Mobile Ad-hoc Networks (MANET). More specifically, we describe the performance behavior of BIC, Cubic, TCP Compound, Vegas, Reno and Westwood congestion control algorithms. The evaluation is simulated through Network Simulator (NS2) and the performance of these congestion control algorithms is analyzed with suitable metrics.
Performance Evaluation of TCP with Adaptive Pacing and LRED in Multihop Wirel...ijwmn
Transmission Control Protocol (TCP) was designed to provide reliable end-to-end delivery of
data over unreliable networks. In practice, most TCP deployments have been carefully designed in the
context of wired networks. Ignoring the properties of wireless and Ad-hoc Networks can lead to TCP
implementations with poor performance. In a wireless network, however packet losses occur more often
due to unreliable wireless links than due to congestion. When using TCP over wireless links, each packet
loss on the wireless link results in congestion control measures being invoked at the source. This causes
severe performance degradation. If there is any packet loss in wireless networks, then the reason for that
has to be found out. If there is congestion, then only congestion control mechanism has to be applied.
This work shows the performance of TCP with Adaptive Pacing (TCP-AP) and Link Random Early
Discard (LRED) as queuing model in multihop transmission when the source and destination nodes are
in mobile nature. The adaptive pacing technique seeks to improve spatial reuse. The LRED technique
seeks to react earlier to link overload. This paper consists of simulated environment results under
different network scenarios. This work proves that the combination of TCP-AP and LRED give much
better result than as the individual technique. Simulations are done with the use of NS-2.
This paper prefers a fuzzy-logic-based sending rate adaption scheme named FSR(Fuzzy Sending Rate) intending to improve the evenness of TCPFriendly Multicast Congestion Control (TFMCC). To mitigate fluctuation of sending rate for TFMCC sender, FSR intends, five actions and link utilization for tuning sending rate and uses a fuzzy controller to determine which operation should be reaped according to the feedback information from CLR (current limiting receiver). Asymmetrical membership functions and biased fuzzy inference rules make FSR as friendly to TCP flows as TFMCC. Simulation results show that FSR has exceptional smoothness and fine TCP Friendliness
EFFICIENT ADAPTATION OF FUZZY CONTROLLER FOR SMOOTH SENDING RATE TO AVOID CON...ijcsit
ABSTRACT
This paper prefers a fuzzy-logic-based sending rate adaption scheme named FSR(Fuzzy Sending Rate) intending to improve the evenness of TCPFriendly Multicast Congestion Control (TFMCC). To mitigate fluctuation of sending rate for TFMCC sender, FSR intends, five actions and link utilization for tuning sending rate and uses a fuzzy controller to determine which operation should be reaped according to the feedback information from CLR (current limiting receiver). Asymmetrical membership functions and biased fuzzy inference rules make FSR as friendly to TCP flows as TFMCC. Simulation results show that FSR has exceptional smoothness and fine TCP Friendliness.
Performing Network Simulators of TCP with E2E Network Model over UMTS NetworksAM Publications,India
Wireless links losses result in poor TCP throughput since losses are perceived as congestion by TCP with the evolution of 3G technologies like Universal Mobile Telecommunication System (UMTS), the usage of TCP has become more popular for a reliable end-to-end (e2e) data delivery. However, TCP was initially designed for wired networks and therefore it suffers performance degradation due to the radio signal getting affected by fading, shadowing and interference. There are many strategies proposed by the research community on how to improve the performance of TCP over wireless links such as introducing link-layer retransmission, explicitly notifying the sender of network conditions or using new variants of TCP. As UMTS network coverage and availability are currently experiencing rapid growth, optimization of various internal components of its wireless network is very important. One of the optimization is the introduction of High Speed Downlink Packet Access (HSDPA). This architecture not only allows higher data rates but also more reliable data transfer by the introduction of Hybrid ARQ (HARQ). With this enhancement to the UMTS network, it becomes vital to see the performance of TCP in such a network. Therefore in this thesis, we try to evaluate two aspects of UMTS networks: first, the impact of HSDPA parameters like scheduling algorithm and RLC/MAC-hs buffer size on overall performance of TCP and second, to study the behaviour of two categories of TCP rate and flow control: loss based and delay based. Our simulation shows that delay based TCP tends to perform better than loss based TCP in our selected scenarios. The simulations are performed using the network simulator NS-2 with an e2e network model for enhanced UMTS (EURANE).
IMPACT OF CONTENTION WINDOW ON CONGESTION CONTROL ALGORITHMS FOR WIRELESS ADH...cscpconf
TCP congestion control mechanism is highly dependent on MAC layer Backoff algorithms that
predict the optimal Contention Window size to increase the TCP performance in wireless adhoc
network. This paper critically examines the impact of Contention Window in TCP congestion
control approaches. The modified TCP congestion control method gives the stability of
congestion window which provides higher throughput and shorter delay than the traditional TCP. Various Backoff algorithms that are used to adjust Contention Window are simulatedusing NS2 along with modified TCP and their performance are analyzed to depict the influence of Contention Window in TCP performance considering the metrics such as throughput, delay, packet loss and end-to-end delay
Improving Performance of TCP in Wireless Environment using TCP-PIDES Editor
Improving the performance of the transmission
control protocol (TCP) in wireless environment has been an
active research area. Main reason behind performance
degradation of TCP is not having ability to detect actual reason
of packet losses in wireless environment. In this paper, we are
providing a simulation results for TCP-P (TCP-Performance).
TCP-P is intelligent protocol in wireless environment which
is able to distinguish actual reasons for packet losses and
applies an appropriate solution to packet loss.
TCP-P deals with main three issues, Congestion in
network, Disconnection in network and random packet losses.
TCP-P consists of Congestion avoidance algorithm and
Disconnection detection algorithm with some changes in TCP
header part. If congestion is occurring in network then
congestion avoidance algorithm is applied. In congestion
avoidance algorithm, TCP-P calculates number of sending
packets and receiving acknowledgements and accordingly set
a sending buffer value, so that it can prevent system from
happening congestion. In disconnection detection algorithm,
TCP-P senses medium continuously to detect a happening
disconnection in network. TCP-P modifies header of TCP
packet so that loss packet can itself notify sender that it is
lost.This paper describes the design of TCP-P, and presents
results from experiments using the NS-2 network simulator.
Results from simulations show that TCP-P is 4% more
efficient than TCP-Tahoe, 5% more efficient than TCP-Vegas,
7% more efficient than TCP-Sack and equally efficient in
performance as of TCP-Reno and TCP-New Reno. But we can
say TCP-P is more efficient than TCP-Reno and TCP-New
Reno since it is able to solve more issues of TCP in wireless
environment.
Performance Evaluation of UDP, DCCP, SCTP and TFRC for Different Traffic Flow...IJECEIAES
The demand for internet applications has increased rapidly. Providing quality of service (QoS) requirements for varied internet application is a challenging task. One important factor that is significantly affected on the QoS service is the transport layer. The transport layer provides end-to-end data transmission across a network. Currently, the most common transport protocols used by internet application are TCP (Transmission Control Protocol) and UDP (User Datagram Protocol). Also, there are recent transport protocols such as DCCP (data congestion control protocol), SCTP (stream congestion transmission protocol), and TFRC (TCP-friendly rate control), which are in the standardization process of Internet Engineering Task Force (IETF). In this paper, we evaluate the performance of UDP, DCCP, SCTP and TFRC protocols for different traffic flows: data transmission, video traffic, and VOIP in wired networks. The performance criteria used for this evaluation include throughput, end to end delay, and packet loss rate. Well-known network simulator NS-2 used to implement the UDP, DCCP, SCTP, and TFRC protocols performance comparison. Based on the simulation results, the performance throughput of SCTP and TFRC is better than UDP. Moreover, DCCP performance is superior SCTP and TFRC in term of end-to-end delay.
Comparative evaluation of bit error rate for different ofdm subcarriers in ra...ijmnct
In the present situation, the expectation about the quality of signals in wireless communication is as high as possible. This quality issue is dependent upon the different communication parameters. One of the most important issues is to reduce the bit error rate (BER) to enhance the performance of the system. This paper provides a comparative analysis on the basis of this bit error rate. I have compared the BER for different number of subcarriers in OFDM system for BPSK modulation scheme. I have taken 6 varieties of data subcarriers to analyze this comparison. Here my target is to reach at the lowest level of BER for BPSK modulation. That is achieved at 2048 number of subcarriers.
A downlink scheduler supporting real time services in LTE cellular networksUniversity of Piraeus
The wide spread of real-time services in wireless networks demands scheduling mechanisms supporting strict Quality of Service (QoS) requirements. Nevertheless, the specifications of the LTE standard for mobile connectivity defined by the 3rd Generation Partnership Project (3GPP) does not impose any specific scheduler for the proper allocation of resources to services. Therefore, several LTE schedulers have been proposed in the literature meeting the QoS requirements of modern services. In this paper a QoS aware scheduler for the LTE downlink is proposed namely the FLS-Advanced (FLSA) aiming at prioritizing real-time traffic. The proposed scheduler has been built on three distinct levels assigning the available radio resources to services according to their requirements. Based on simulation results, the FLSA outperforms in terms of packet loss ratio, attainable throughput and fairness the performance of existing schedulers including PF, MLWDF, EXP/PF, FLS, EXP RULE and LOG RULE.
IJERA (International journal of Engineering Research and Applications) is International online, ... peer reviewed journal. For more detail or submit your article, please visit www.ijera.com
Unit 8 - Information and Communication Technology (Paper I).pdfThiyagu K
This slides describes the basic concepts of ICT, basics of Email, Emerging Technology and Digital Initiatives in Education. This presentations aligns with the UGC Paper I syllabus.
Instructions for Submissions thorugh G- Classroom.pptxJheel Barad
This presentation provides a briefing on how to upload submissions and documents in Google Classroom. It was prepared as part of an orientation for new Sainik School in-service teacher trainees. As a training officer, my goal is to ensure that you are comfortable and proficient with this essential tool for managing assignments and fostering student engagement.
Welcome to TechSoup New Member Orientation and Q&A (May 2024).pdfTechSoup
In this webinar you will learn how your organization can access TechSoup's wide variety of product discount and donation programs. From hardware to software, we'll give you a tour of the tools available to help your nonprofit with productivity, collaboration, financial management, donor tracking, security, and more.
Biological screening of herbal drugs: Introduction and Need for
Phyto-Pharmacological Screening, New Strategies for evaluating
Natural Products, In vitro evaluation techniques for Antioxidants, Antimicrobial and Anticancer drugs. In vivo evaluation techniques
for Anti-inflammatory, Antiulcer, Anticancer, Wound healing, Antidiabetic, Hepatoprotective, Cardio protective, Diuretics and
Antifertility, Toxicity studies as per OECD guidelines
June 3, 2024 Anti-Semitism Letter Sent to MIT President Kornbluth and MIT Cor...Levi Shapiro
Letter from the Congress of the United States regarding Anti-Semitism sent June 3rd to MIT President Sally Kornbluth, MIT Corp Chair, Mark Gorenberg
Dear Dr. Kornbluth and Mr. Gorenberg,
The US House of Representatives is deeply concerned by ongoing and pervasive acts of antisemitic
harassment and intimidation at the Massachusetts Institute of Technology (MIT). Failing to act decisively to ensure a safe learning environment for all students would be a grave dereliction of your responsibilities as President of MIT and Chair of the MIT Corporation.
This Congress will not stand idly by and allow an environment hostile to Jewish students to persist. The House believes that your institution is in violation of Title VI of the Civil Rights Act, and the inability or
unwillingness to rectify this violation through action requires accountability.
Postsecondary education is a unique opportunity for students to learn and have their ideas and beliefs challenged. However, universities receiving hundreds of millions of federal funds annually have denied
students that opportunity and have been hijacked to become venues for the promotion of terrorism, antisemitic harassment and intimidation, unlawful encampments, and in some cases, assaults and riots.
The House of Representatives will not countenance the use of federal funds to indoctrinate students into hateful, antisemitic, anti-American supporters of terrorism. Investigations into campus antisemitism by the Committee on Education and the Workforce and the Committee on Ways and Means have been expanded into a Congress-wide probe across all relevant jurisdictions to address this national crisis. The undersigned Committees will conduct oversight into the use of federal funds at MIT and its learning environment under authorities granted to each Committee.
• The Committee on Education and the Workforce has been investigating your institution since December 7, 2023. The Committee has broad jurisdiction over postsecondary education, including its compliance with Title VI of the Civil Rights Act, campus safety concerns over disruptions to the learning environment, and the awarding of federal student aid under the Higher Education Act.
• The Committee on Oversight and Accountability is investigating the sources of funding and other support flowing to groups espousing pro-Hamas propaganda and engaged in antisemitic harassment and intimidation of students. The Committee on Oversight and Accountability is the principal oversight committee of the US House of Representatives and has broad authority to investigate “any matter” at “any time” under House Rule X.
• The Committee on Ways and Means has been investigating several universities since November 15, 2023, when the Committee held a hearing entitled From Ivory Towers to Dark Corners: Investigating the Nexus Between Antisemitism, Tax-Exempt Universities, and Terror Financing. The Committee followed the hearing with letters to those institutions on January 10, 202
June 3, 2024 Anti-Semitism Letter Sent to MIT President Kornbluth and MIT Cor...
ETFRC: Enhanced TFRC for Media Traffic over Internet
1. Mohammad A. Talaat, Magdi A. Koutb & Hoda S. Sorour
International Journal of Computer Networks (IJCN), Volume (3) : Issue (3) : 2011 167
ETFRC: Enhanced TFRC for Media Traffic over Internet
Mohammad A. Talaat moadly@tedata.net.eg
Faculty of Electronic Engineering
University of Menofiya
Menouf, Egypt
Magdi A. Koutb magdykoutb@yahoo.com
Faculty of Electronic Engineering
University of Menofiya
Menouf, Egypt
Hoda S. Sorour hodasorour@yahoo.com
Faculty of Electronic Engineering
University of Menofiya
Menouf, Egypt
Abstract
The evident increase in media traffic over Internet is expected to worsen its congestion state.
TCP-friendly rate control protocol TFRC is one of the most promising congestion control
techniques developed so far. TFRC has been thoroughly tested in terms of being TCP-friendly,
responsive, and fair. Yet, its impact on the visual quality and the peak signal-to- noise ratio PSNR
of the media traffic traversing Internet is still questionable. In this paper we aimed to point out the
enhancements required for TFRC that enables producing the maximum PSNR value for Internet
media traffic. Firstly, we suspected the default value of n that represents the number of loss
intervals used in calculating the loss event rate in the TFRC equation. This value is
recommended to be set to 8 according to the latest RFC of TFRC. We investigated the effect of
modifying the TFRC mechanism on the resulting PSNR of the transmitted video over Internet
using TFRC via switching n across the values from 2 to 16. We investigated the effect of such
variation over a simulated network environment to study its effect on the resulting PSNR for a
number of arbitrary video sequences. Our simulations results showed that running TFRC with
n=11 led to reaching the maximum PSNR values among all the examined values of n including its
default value. Secondly, we tested the impact on the PSNR of another modification in the TFRC
mechanism via switching both values of n and Nfb which is frequency of feedback messages sent
by TFRC receiver to its sender every round-trip time RTT. The default value of Nfb is 1; hence we
scanned every possible combination of n and Nfb ranging from 2 to 16, and from 1 to 4,
respectively and recorded the produced PSNR. It was obvious that several other combinations of
n and Nfb produced higher PSNR values other than their default values in the request for
comment RFC of TFRC. We hereby suggest using an enhanced TFRC that we abbreviated as
ETFRC which has the values of n and Nfb value set to 4 and 11 respectively as a replacement for
the traditional TFRC to enable reaching higher PSNR for media traffic over Internet.
Keywords: Congestion Control, TFRC, PSNR, Media Traffic.
1. INTRODUCTION
The percentage of media traffic traversing Internet has remarkably increased over the last
decade. Applications pushing media traffic such as video on demand VoD, video conferencing,
and various video streaming websites have been lately invading the cyber space. The best-effort
existing IP infrastructure was not primarily designed to suffice the quality of service QoS
requirements of such traffic. Both of the current UDP and TCP have drawbacks when used as the
transport protocol for media traffic. TCP seems to break the delay constraints due to its
2. Mohammad A. Talaat, Magdi A. Koutb & Hoda S. Sorour
International Journal of Computer Networks (IJCN), Volume (3) : Issue (3) : 2011 168
acknowledgments; meanwhile UDP shows aggressiveness in acquiring the available bandwidth
to accomplish the streaming task. UDP leaves an unfair share for the co-existing TCP flows which
leads to congestion status.
During the periods of congestion; routers tend to discard legitimate packets traversing a certain
bottleneck to be able to serve the aggressive media packets. Efforts have been made by
researchers to control congestion; they tried to balance between allowing for QoS achievement
and acting in a TCP-friendly manner at the same time.
This was via building congestion control protocols that leave a fair share of bandwidth for the
concurrent TCP flows traversing across the same bottleneck. Protocols designed for this purpose
have been tested regarding compatibility with the TCP-friendliness concept defined in [1].
TFRC presented in [2] was one of the congestion control protocols that managed to achieve
remarkable smoothness in the variations of its rate of transmission in addition to fulfilling the TCP-
friendliness conditions. For this reason TFRC was the best current promising candidate for media
streaming applications, where its smoothness helps in reducing the undesired jitter of the
perceived video. TFRC has been extensively tested in terms of fairness, aggressiveness, and
responsiveness as in [3] and in terms of user-perceived media quality on analytical basis in [4].
Testing the visual quality of the media traffic running over TFRC in terms of its produced PSNR
was made in our previous work in [5]
This paper aims to reach an enhanced version of TFRC named as ETFRC that manages to
produce higher PSNR values than for media traffic over Internet. To achieve this goal a network
simulation topology was built to stream a variety of arbitrary video sequences over TFRC. The n
parameter that represents the number of loss intervals samples used in calculating the loss event
rate in the TFRC equation was switched among different values aiming to determine its optimum
value for the best visual quality of video sequences transmitted over TFRC. This quality is
measured in terms of the produced RSNR values for the streamed videos. The optimal value for
n was found to be “eleven” where the maximum PSNR values were observed. Another
investigation was made through switching the values of both of n along with Nfb which is the
frequency of feedback messages per RTT. We aimed to figure out the combination of n and Nfb
that leads to producing the maximum PSNR values for the video sequences traversing Internet
using TFRC. Hence, TFRC is suggested to enhance its mechanism to be ETFRC that uses n=11
and Nfb=4 instead of their default values used in traditional TFRC.
The rest of this paper is organized as follows: Section II covers the TFRC literature and its
modified versions tackling the media streaming task over the last decade. Section III explains the
enhancement we proposed to TFRC mechanism to produce the proposed ETFRC for media
traffic over Internet. Section IV explains our simulation environment, the tool-set used in
simulations, the topology of the simulated network, the link parameters set, and the
characteristics of the running video sequences. Section IV presents the simulations output
focusing on the PSNR values for the tested TFRC in order to show that the enhancement
introduced to reach our proposed ETFRC managed to produce the maximum PSNR values.
Finally Section V concludes this work.
2. THE TFRC PROTOCOL
2.1 TCP Friendly Congestion Control
The TCP friendly congestion control schemes lies into two main categories according to [1]: (i)
single rate schemes and (ii) multi-rate schemes. Unicast applications tend to utilize the single rate
protocols where all recipients receive data with the same rate. This feature limits the scalability of
the protocol towards bandwidth variations that exist in the path to some recipients. Multi-rate
protocols are more flexible and enable the allocation of different rates for different recipients
which makes it more appropriate for the multicast applications.
3. Mohammad A. Talaat, Magdi A. Koutb & Hoda S. Sorour
International Journal of Computer Networks (IJCN), Volume (3) : Issue (3) : 2011 169
Each of those two top categories can be sub-divided into other two sub-categories as follows: (i)
rate-based schemes and (ii) window-based schemes.
Some of the rate-based schemes apply the additive increase multiplicative decrease AIMD
approach embedded in TCP. Other rate-based just tune their sending rate in accordance with a
TCP model. In both cases the reliability feature of TCP is absent. Example protocols that lie in
this category are RAP [6], LDA+ [7], and TFRC.
TCP itself is a window-based protocol, yet some problems should be considered when applying
this mechanism on multicast connections. Multicast TCP MTCP [8] is an example protocol of this
multicast category that managed to deal with these problems.
2.2 TFRC Protocol
TFRC is the evolution of TFRCP [9]. It was mainly developed for unicast communications but it
can be adapted for multicast. Its sending rate is tuned according to the TCP complex equation
(1).
Where the parameters are as follows:
• T: TCP throughput
• RTT: Round-trip time
• RTO: Retransmission time-out value
• S: Segment size
• P: Rate of packet loss
• b: Number of packets acknowledged by each ACK
• : Maximum congestion window size cwnd.
TFRC uses its sophisticated mechanisms to gather the equation parameters. The average loss
interval is the chosen method to fulfill the requirements of the loss rate estimation. The loss rate is
measured utilizing the latest 8 loss intervals through tracking the number of packets between
consecutive loss events. The default number of loss intervals n utilized is recommended to be set
to 8 according to the TFRC latest RFC which has the number 5348. The average of a specified
number of loss intervals is calculated using decaying weights so that old loss intervals
contribution in this average is less. The loss rate is considered as the inverse of the average loss
interval size.
Some additional mechanisms are adopted to prevent TFRC from responding aggressively to
single loss events, and to guarantee that the sending rate adapts quickly to the long intervals that
are loss-free. RTT is measured by sending feedback time stamps to sender.
TFRC goes through a slow-start phase directly after starting just like TCP in order to increase its
rate to reach a fair share of bandwidth. The slow-start phase is ended by reporting a loss event.
TFRC receiver updates the equation parameters and feeds them back to sender to adjust its rate
every round-trip time RTT. Hence a feedback message is sent once per RTT which means that
the default Nfb=1, leading to the recalculation of the sending rate for TFRC only once per RTT.
TFRC adopts additional delay-based congestion avoidance by adjusting the inter-packet gap
which would be applied in some environments that do not support the TCP complex equation.
TFRC main advantage is its stable and smooth sending rate variations. This feature fits in the
media application transmission besides being responsive to the co-existing traffic in a TCP-
friendly manner.
4. Mohammad A. Talaat, Magdi A. Koutb & Hoda S. Sorour
International Journal of Computer Networks (IJCN), Volume (3) : Issue (3) : 2011 170
2.3 TFRC Enhancement Attempts
Several attempts were made to enhance the performance of TFRC in order to suit the media
traffic requirements such as what discussed in the following lines:
In [10] authors observed some performance degradation for TFRC over wireless networks, and
hence they tried to customize it through a more advanced equation. This equation was reached
via modeling wireless TCP rather than wired TCP. Applying this equation led to a remarkable
throughput increase with about 30% over wireless networks with loss of 10% while maintaining
the TFRC main features of TCP-friendliness and smoothness.
In [11] an attempt was made to enhance TFRC performance over mobile and pervasive networks
that focused on overcoming data losses due to the frequent loss of connectivity. The method
used was applying a mechanism that resembles Freeze–TCP when a disconnection incident is
expected. Additionally, a probing mechanism to enable speedy adaptation to new network
conditions was proposed.
Another enhancement was made in [12] where authors tackled the problem of keeping fairness
and smoothness of TFRC media streams when existing among other streams. They proposed
MulTFRC that was successful in keeping low delay values to satisfy the media QoS
requirements.
In [13] the enhancement made was that authors computed the rate gap between the ideal TCP
throughput and the smoothed TFRC throughput replacing it. Any rate gain from this gap was
opportunistically exploited via video encoding. A frame complexity measure is specified to
determine the additional rate to be used from this rate gap, and then the target rate for the
encoder and the final sending rate are negotiated through the same frame of complexity.
In [14] authors suggested incorporating the selective retransmission concept into TFRC.
Retransmission of lost packets is done selectively when no congestion case is present. Selective
retransmission was shown to have a significant positive effect on the streamed media quality over
TFRC.
In [15] authors claimed that TFRC does not perform satisfactorily on multi-hop ad-hoc wireless
networks. They saw that TFRC sending rate can be deceived by MAC layer contention effects
such as retransmission and exponential back-off. Hence, they proposed enhancing TFRC by
introducing RE-TFRC. RE-TFRC used measurements of the current round-trip time and a model
of wireless delay to prevent TFRC from overloading the MAC layer while keeping its TCP-
friendliness feature.
In [16] a performance analysis of a QoS-aware congestion control mechanism named guaranteed
TFRC (gTFRC) was presented. gTFRC was embedded into the enhancement transport protocol
(ETP) that enables protocol mechanisms to be dynamically controlled. gTFRC managed to reach
a minimum guaranteed transfer rate for any given round-trip values and any network provisioning
conditions as well
In [17] an extension for TFRC was suggested in order to support variable packet size streams.
Variable packet size is already utilized over Internet in both video and voice over IP VoIP
transmission. This enhancement of TFRC was made through a modified concept of TCP-
friendliness and was validated to perform better than the original TFRC with the packets of
variable sizes.
In [18] authors proposed an enhanced TFRC based on step explicit congestion notification ECN
making. They found that TFRC has poor performance over wireless networks because it
accounts for wireless losses as congestion. They managed to reach higher throughput via
utilizing the appropriate loss differentiation and congestion notification.Their proposed enhanced
5. Mohammad A. Talaat, Magdi A. Koutb & Hoda S. Sorour
International Journal of Computer Networks (IJCN), Volume (3) : Issue (3) : 2011 171
TFRC kept reasonable friendliness to the co-existing TCP flows according to their results.
The above enhancement attempts focused on achieving a better quality of data transmitted over
TFRC while maintaining its main advantageous characteristics of being TCP- friendly and fair in
acquiring a bandwidth share.
3 ENHANCED TFRC ETFRC PROTOCOL
The enhancement that we introduced to TFRC to produce ETFRC was increasing the number of
sample loss intervals n used to calculate the rate of packet loss P from n=8 to n=11 We also
studied the effect of combining the different values of n along with other values of feedback
frequency represented by Nfb parameter. The effect of varying the Nfb solely was studied in [19].
Our goal is to demonstrate through simulations that combining the values of n=11 and Nfb=4
leads to the maximum PSNR values for the transmitted video over TFRC. For the sake of
completeness, other effects of such increase on the TFRC mechanism of work should be
discussed. In [20] authors discussed the impact of increasing the feedback frequency from
different perspectives that we summarize here:
3.1 The Impact of Increasing Nfb on TFRC Mechanism
To have a better view for the background image of TFRC, we have to understand that there are
two key parameters that drive the TFRC mechanism in Eq.1 which are the loss rate p and the
experienced RTT. In our work we used the value of Nfb=1 as a reference value as this frequency
is considered as the default of TFRC. Simulations made in [20] showed that feedback frequencies
greater than one per RTT lead to lower drop rate. It was also clear that the feedback frequency is
not correlated to P, where both values were randomly distributed. Hence, Nfb has no critical
effect on P from the macroscopic perspective.
The analysis made in [20] for TFRC demonstrated the impact of increasing the feedback
frequency on RTT, it showed that during the slow-start of TFRC the link suffers from under-
utilization as well as slow convergence. For the first hundred seconds the value of RTT in TFRC
equals to that of the link propagation delay, meanwhile after this period, TFRC enters the steady
state where the increase in feedback frequencies causes the experienced RTT to increase.
3.2 The Impact of Increasing Nfb on Sending Rate
Researches also noted that when the feedback frequency increases the rate of the senders tend
to decrease, which seems to contradict with what was expected of decrease in network
congestion and the decrease in RTT consequently. The higher feedback frequency was found to
improve the accuracy of the estimated RTT and the accuracy of the estimated p in consequent.
This enables TFRC to acquire the network resources with the minimum required sending rate due
to the fact that the more accurate the sense of the network congestion parameter the faster the
adaptation of TFRC to the network conditions.
3.3 The Impact of Increasing Nfb on Network Parameters
Increasing the feedback frequency of TFRC from one to four had a positive impact on its
responsiveness. However the resulting fairness depended on how lost packets were distributed
among flows. The link utilization also followed the same behavior of increase like that of
responsiveness when having multiple feedbacks per RTT
4 SIMULATIONS ENVIRONMENT
This section describes our simulation environment used to perform the PSNR evaluation of the
media traffic over TFRC. The main three components of these simulations are the tool-set used,
the topology of the simulated network created using ns-2.30 [21], and the group of video
sequences arbitrarily chosen for this purpose.
4.1 Evalvid Tool-set and Evalvid-RA
In [22] Chih-heng Ke et al. proposed a novel and complete tool-set for evaluating the quality of
MPEG video delivery over simulated networks environment. This tool-set is based on the EvalVid
6. Mohammad A. Talaat, Magdi A. Koutb & Hoda S. Sorour
International Journal of Computer Networks (IJCN), Volume (3) : Issue (3) : 2011 172
framework [23]. They managed to let ns-2 as a general network simulator replace the EvalVid
simple error simulation model through extending its connecting interfaces. This allowed
researchers and practitioners in general to simulate and analyze the performance of real video
streams with consideration for the video semantics under a vast range of network scenarios.
The tool-set valuable feature is that it allows for the examination of the relationship between two
well-known objective metrics for QoS assessment of video quality of delivery which are the PSNR
and the fraction of decodable frames.
As shown in Fig. 1 the concept of this tool-set is built upon creating trace files from an encoded
raw video. These files are text files and are fed into the simulation environment to be used as
traffic generators based on the encoded video parameters. The output of the simulation process
can be decoded as well to produce the output video file of the simulation. The quality of the
output file and the original video file can then be compared to obtain a representative PSNR value
for the media quality of the produced video from simulation
FIGURE 1: EvalVid Tool-set.
EvalVid-RA proposed in [24] is an extension of EvalVid tool-set that supports the rate adaptive
media content as well as the TFRC protocol in the simulation environment, thus it was chosen to
be used in this work.
4.2 The Simulations Network Topology
Our simulation topology in this paper is the same simple topology used in [24]. As shown in Fig.
2, it is a simple dumbbell topology composed of four traffic sources and four traffic destinations.
We used ns-2.30 as our network simulator to build this topology. Both S0 and S1 are fed with the
simulated video trace files while S2 and S3 generate TCP traffic.
The video traffic is shaped variable bit-rate traffic rate adaptive (RA-SVBR), and the bottleneck
between the routers R0 and R1 is 40Mbit/s link with propagation delay of 10ms. The access
network capacities were set to 32Mbit/s with 5ms delay producing a one-way propagation delay
of 20ms. The fair share after starting all sources was over 625Kbit/s and both R0 and R1 routers
used ordinary random early detection (RED).
7. Mohammad A. Talaat, Magdi A. Koutb & Hoda S. Sorour
International Journal of Computer Networks (IJCN), Volume (3) : Issue (3) : 2011 173
Using ns-2.30 enabled testing TFRC with different Nfb values to find optimum value for ETFRC
FIGURE 2: Simulations Topology
4.3 The Video Sequences Used in Simulations
Two video sequences used in our simulations as shown in Table 1. They are provided for
research purposes by Arizona State University research group in [25].
(2)
Where k and MSE are as follows:
k: the number of bits per pixel
MSE: the mean square error of the luminance component.
They were used in YUV format where they are firstly encoded using ffmpeg.exe and then passed
by the mp4.exe which are both component files of the used tool-set.
A brief description for each of the videos content is presented as follows:
1. Bridge-close: a close scene of Charles bridge
2. Mobile: panning of moving toys
The following table shows the video sequences used and their number of frames and complexity
of motion:
TABLE 1: The Video Sequences Used in Simulations.
The final output of the tool-set is a text file that contains a table of two columns where the PSNR
value of each compared video frame calculated according to equation (2) .is recorded in decibels
(dB) corresponding to their frame numbers.
Video Sequence No. of Frames Motion Complexity
Bridge-close 2001 Low
Mobile 300 Medium
S0
R0
S1
S2
S3
R1
D0
D1
D2
D3
RA-SVBR
RA-SVBR
TCP
TCP
8. Mohammad A. Talaat, Magdi A. Koutb & Hoda S. Sorour
International Journal of Computer Networks (IJCN), Volume (3) : Issue (3) : 2011 174
We utilized the mean PSNR value for each simulation video file for our evaluation purpose to be
compared with the reference mean PSNR produced by TFRC having the default Nfb value of
one. The resulting text files are then fed into the ns-2 simulation file where the output is
concatenated through et_ra.exe tool and then decoded. The decoded file quality is the compared
to the original file quality using the psnr.exe program as demonstrated in Fig. 1.
5 SIMULATIONS RESULTS
The goal of building and running the simulated environment explained above was to investigate
the effect of switching the n and Nfb parameter over a range of values on the resulting PSNR
values of the output files.
FIGURE 3: PSNR Values Vs n
The goal of building and running the simulated environment explained above was to investigate
the effect of switching the n and Nfb parameter over a range of values on the resulting PSNR
values of the output files.
Our results are based on calculating the mean PSNR values for each video sequence which are
the results of comparing the output video files of simulation over TFRC protocol and the original
video files. Those PSNR values were calculated using the psnr.exe program as a part of the
EvalVid tool-set. The program compares each output video file to the original file on frame-by-
frame basis. It produces a text file that contains a PSNR value for each of the video frames. The
mean of those PSNR values were computed to be compared to those of the default PSNR values
of the TFRC
Knowing that the default value of n in TFRC is “8” and of Nfb in TFRC is “1”, we managed to
switch n over a range of values from 2 to 16 and Nfb over the values of 1, 2, 3, and 4.
Our first finding was that the PSNR values of the output files all lie in the acceptable range at the
chosen values of n and Nfb. This means that TFRC is a suitable candidate for running the media
traffic from both points of views of TCP-friendliness and media quality maintenance. The output
files were visually meaningful. They had a degraded quality with respect to the original video files,
but all the files were considered to be visually acceptable by human eye as well as PSNR values.
Our next goal in this paper was specifying the exact combination of values of both n and Nfb that
produces the maximum PSNR for the output video files, and that would be set as the defaults in
the suggested ETFRC. To achieve this goal we recorded the PSNR for each of the values chosen
for n and Nfb. Fig. 3 represents the PSNR values of both simulated videos versus the range of
values scanned for n while Fig. 4 represents the PSNR values of one of the videos versus all the
possible combinations of n and Nfb over their scanned ranges for testing.
9. Mohammad A. Talaat, Magdi A. Koutb & Hoda S. Sorour
International Journal of Computer Networks (IJCN), Volume (3) : Issue (3) : 2011 175
We can also note in Fig. 3 that n=11 leads to the maximum PSNR values among the other tested
values. While in Fig. 4 we can notice that the combination of n=11 and Nfb=4 is the optimum for
having maximum PSNR for the tested video. We believe that our simulation results are helpful for
measuring the performance of the TFRC from a point of view that has not been thoroughly
tackled before which is the PSNR of videos running over it.
FIGURE 4: PSNR Values Vs n and Nfb
6 CONCLUSION
The problem of Internet congestion control has been handled by researchers over the last
decade. It was observed that the quantity of media traffic traversing the Internet has
tremendously increased due to the increase in the number of emerging applications running such
traffic which led to worsening the case of congestion.
Several congestion control protocols have been developed to face the problem and also have
been tested from the TCP-friendliness point of view and achieved promising performance in many
cases, but the media traffic currently booming over Internet imposed some additional criteria on
the congestion control protocols other than just being TCP-friendly and fair in bandwidth
acquiring. These criteria focused on delivering the media traffic with an acceptable PSNR quality
leading to a visually meaningful and acceptable traffic.
TFRC according to researches is a good candidate protocol for this target. It can balance
between accomplishing the TCP-friendliness task and allowing for some QoS constraints to be
met. Several researches pointed out that TFRC with its current mechanism of work is not the
ideal protocol for congestion handling, and hence many enhancement attempts were made to
reach a more suitable form of it.
A number of researches targeted the evaluation of TFRC in terms of TCP-friendliness and
fairness and many tests were done for this purpose either in the simulated environments or in the
real-world. TFRC has also been tested so far regarding the quality of the media traffic running
over it.
This work managed to find an enhanced version of TFRC named as ETFRC through changing
the default values of both parameters n and Nfb in TFRC from eight to eleven and from one four
respectively to form the ETFRC. This was by testing the quality of a group of videos when
10. Mohammad A. Talaat, Magdi A. Koutb & Hoda S. Sorour
International Journal of Computer Networks (IJCN), Volume (3) : Issue (3) : 2011 176
transmitted over TFRC while switching the values of n and Nfb over a range of nominated
integers. This testing utilized the simulation environment and was made in terms of the visual
usefulness of the received video files. It was also made in terms of the mean PSNR values in dB
of the output files of the simulation process when compared to the originally transmitted files.
TFRC was shown to produce acceptable quality for the received video files. This emphasizes the
fact that TFRC is still the candidate for the congestion control problem solving. It was also shown
through the simulations results that the performance of TFRC in terms of quality was enhanced
slightly with the above suggested changing in defaults
We hereby propose the ETFRC protocol as an enhancement for TFRC where the number of
sample loss intervals used is eleven instead of eight and the feedback frequency utilized value is
four instead of one. We also believe that this work can be helpful for researchers handling the
congestion control problem and researchers trying to enhance the performance of TFRC in order
to increase its capabilities of being the congestion control problem solution.
7 REFERENCES
[1] J. Widmer, R. Denda, and M. Mauve, “A survey on tcp-friendly congestion control
(extended version),” Technical Report, Department for Mathematics and Computer
Science, University of Mannheim, Feb. 2001.
[2] S. Floyd, M. Handley, J. Padhye, and J. Widmer, “Equation-based congestion control for
unicast applications,” in Proc. ACM SIGCOMM, Stockholm, Sweden, Aug. 2000, pp. 43 –
56.
[3] S. Tsao, Y. Lai, and Y. Lin, “Taxonomy and Evaluation of TCP-Friendly Congestion Control
Schemes on Fairness, Aggressiveness, and Responsiveness.” IEEE Network, vol. 21, no.
6, 2007, pp. 6-15.
[4] Lisong Xu and Josh Helzer, “Media Streaming via TFRC: an Analytical Study of the Impact
of TFRC on User-perceived Media Quality,” Computer Networks, vol. 51, no. 17, 2007, pp.
4744-4764.
[5] M. A. Talaat, M. A. Koutb, and H. S. Sorour, "PSNR Evaluation for Media Traffic over
TFRC," International Journal of Computer Networks and Communications, vol. 1, no. 3, pp.
71-76, October 2009.
[6] R. Rejaie, M. Handley, and D. Estrin, “Rap: An end-to-end rate-based congestion control
mechanism for real-time streams in the internet,” in Proc. IEEE Infocom, Mar. 1999.
[7] D. Sisalem and A. Wolisz, “Lda+ tcp-friendly adaptation: A measurement and comparison
study,” in Proc. International Workshop on Network and Operating Systems Support for
Digital Audio and Video (NOSSDAV), June 2000.
[8] I. Rhee, N. Balaguru, and G. Rouskas, “Mtcp: scalable tcp-like congestion control for
reliable multicast,” in Proc. of IEEE INFOCOM, March 1999, vol.3, pp. 1265-1273.
[9] J. Padhye, D. Kurose, and R. Towsley, “A model based tcp-friendly rate control protocol,” in
Proc. International Workshop on Network and Operating Systems Support for Digital Audio
and Video (NOSSDAV), June 1999.
[10] B. Zhou, C. P. Fu, V. O. K. Li, "Tfrc veno: an enhancement of TCP-friendly rate control over
wired/wireless networks,"in ProcIEEE ICNP, Oct. 2007.
11. Mohammad A. Talaat, Magdi A. Koutb & Hoda S. Sorour
International Journal of Computer Networks (IJCN), Volume (3) : Issue (3) : 2011 177
[11] O. Mehani and R. Boreli, "Adapting TFRC to mobile networks with frequent
disconnections," in Proc. CoNEXT 2008, 4th ACM International Conference on emerging
Networking EXperiments and Technologies, K. W. Ross and L. Tassiulas, Eds. New York,
NY, USA: ACM, 2008.
[12] D. Darjanovic and Michael Welzl, “Multfrc: providing weighted fairness for multimedia
applications (and others too!).” in Proc. ACM Computer Communication Review, 2009.
[13] E. Tan, J. Chen, S. Ardon, and E. Lochin, “Video tfrc,” IEEE International Conference on
Communications, Beijing, China, 2008.
[14] A. Huszak and S. Imre, “Tfrc-based selective retransmission for multimedia applications,” in
Proc. Advances in Mobile Multimedia, Jakarta, Indonesia, 2007, pp. 53-64.
[15] M. Li, C. Lee, E. Agu, M. Claypool, and R. Kinicki, “Performance enhancement of TFRC in
wireless ad-hoc networks,” in Proc of. 10th
International Conference on Distributed
Multimedia Systems (DMS), California, USA, September 2004.
[16] G. Jourjon, E. Lochin, L. Dairaine, P. Senac, T. Moors, and A. Seneviratne,
“Implementation and performance analysis of a QoS-aware TFRC mechanism," in Proc. of
IEEE ICON, Singapore, September 2006.
[17] P. G. Vasallo, “Variable packet size equation-based congestion control,” International
Computer Science Institute ICSI, Technical Report, April, 2000.
[18] Zhuonong Xu, Jiawei Huang, and Jianxin Wang, Jin Ye, "An enhanced tfrc protocol based
on step ecn marking," International Conference on Communications and Mobile Computing
(CMC), Shenzen, China, April 2010.
[19] Mohammad A. Talaat, Magdi A. Koutb, and Hoda S. Sorour, “Etfrc: enhanced tfrc for media
traffic,” International Journal of Computer Applications (IJCA), vol. 18, no. 6, pp. 1-8, March
2011.
[20] Dino M. Lopez-Pacheco, Emmanuel Lochin, Golam Sarwar, and Roksana Boreli,
"Understanding the impact of tfrc feedbacks frequency over long delay links," Global
Information Infrastructure Symposium (IEEE GIIS 2009), Hammamet, Tunisia, 2009.
[21] University of California Berkeley, “The Network Simulator - ns-2,” [Online]
http://www.isi.edu/nsnam/ns.
[22] C. H. Ke, C. K. Shieh, W. S. Hwang, A. Ziviani, “An Evaluation Framework for More
Realistic Simulations of MPEG Video Transmission”, Journal of Information Science and
Engineering, vol. 24, no. 2, pp.425-440, March 2008.
[23] J. Klaue, B. Rathke, and A. Wolisz, "Evalvid - A framework for video transmission and
quality evaluation,” in Proc. 13th International Conference on Modeling Techniques and
Tools for Computer Performance Evaluation, pp. 255-272, Urbana, Illinois, USA,
September 2003.
[24] A. Lie, and J. Klaue "Evalvid-RA: Trace Driven Simulation of Rate Adaptive MPEG-4 VBR
Video", Multimedia Systems, vol. 14, no. 1, 13. November 2007.
[25] Arizona State University, “YUV Video Sequences” [Online]
http://trace.eas.asu.edu/yuv/index.html.