The document proposes a system for distributed video streaming from multiple senders to a single receiver. It presents a protocol that allows the receiver to calculate optimal sending rates for each sender based on estimated bandwidth and packet loss rates. Each sender then partitions packets to minimize the probability of late arrivals based on its assigned sending rate and estimated delay to the receiver. Simulations showed the system can smooth video delivery across multiple sources when bandwidth from individual senders may vary.
Transmission Control Protocol (TCP) is a fundamental protocol of the Internet Protocol Suite. TCP complements the Internet Protocol (IP), therefore it is common to refer to the internet protocol suit as TCP/IP. TCP is used for error detection, detection of packet loss or out of order delivery of data. TCP requests retransmission, rearranges data and helps with network congestion.
Several congestion control algorithms have been developed, over the last years, to improve TCP's performance over various technologies and network conditions.
The purpose of this assignment is to present TCP, network congestion, congestion algorithms and simulate different algorithms in different network conditions to measure their performance. For this assignment's needs, OPNET IT Guru Academic Edition software was used to accomplish the reproduction of projects that have been already published and gave the wanted results.
Comparison of TCP congestion control mechanisms Tahoe, Newreno and VegasIOSR Journals
The widely used reliable transport protocol TCP, is an end to end protocol designed for the wireline
networks characterized by negligible random packet losses. This paper represents exploratory study of TCP
congestion control principles and mechanisms. Modern implementations of TCP contain four intertwined
algorithms: slow start, congestion avoidance, fast retransmit, and fast recovery. In addition to the standard
algorithms used in common implementations of TCP, this paper also describes some of the more common
proposals developed by researchers over the years. We also study, through extensive simulations, the
performance characteristics of four representative TCP schemes, namely TCP Tahoe, New Reno and Vegas
under the network conditions of bottleneck link capacities for wired network
Connection Establishment & Flow and Congestion ControlAdeel Rasheed
On these slides i describe all the detail about Connection Establishment & Flow and Congestion Control. For more detail visit: https://chauhantricks.blogspot.com/
Computer networks have experienced an explosive growth over the past few years and with
that growth have come severe congestion problems. For example, it is now common to see
internet gateways drop 10% of the incoming packets because of local buffer overflows.
Our investigation of some of these problems has shown that much of the cause lies in
transport protocol implementations (
not
in the protocols themselves): The ‘obvious’ ways
to implement a window-based transport protocol can result in exactly the wrong behavior
in response to network congestion. We give examples of ‘wrong’ behavior and describe
some simple algorithms that can be used to make right things happen. The algorithms are
rooted in the idea of achieving network stability by forcing the transport connection to obey
a ‘packet conservation’ principle. We show how the algorithms derive from this principle
and what effect they have on traffic over congested networks.
In October of ’86, the Internet had the first of what became a series of ‘congestion col-
lapses’. During this period, the data throughput from LBL to UC Berkeley (sites separated
by 400 yards and two IMP hops) dropped from 32 Kbps to 40 bps. We were fascinated by
this sudden factor-of-thousand drop in bandwidth and embarked on an investigation of why
things had gotten so bad. In particular, we wondered if the 4.3
BSD
(Berkeley U
NIX
)
TCP
was mis-behaving or if it could be tuned to work better under abysmal network conditions.
The answer to both of these questions was “yes”.
Transmission Control Protocol (TCP) is a fundamental protocol of the Internet Protocol Suite. TCP complements the Internet Protocol (IP), therefore it is common to refer to the internet protocol suit as TCP/IP. TCP is used for error detection, detection of packet loss or out of order delivery of data. TCP requests retransmission, rearranges data and helps with network congestion.
Several congestion control algorithms have been developed, over the last years, to improve TCP's performance over various technologies and network conditions.
The purpose of this assignment is to present TCP, network congestion, congestion algorithms and simulate different algorithms in different network conditions to measure their performance. For this assignment's needs, OPNET IT Guru Academic Edition software was used to accomplish the reproduction of projects that have been already published and gave the wanted results.
Comparison of TCP congestion control mechanisms Tahoe, Newreno and VegasIOSR Journals
The widely used reliable transport protocol TCP, is an end to end protocol designed for the wireline
networks characterized by negligible random packet losses. This paper represents exploratory study of TCP
congestion control principles and mechanisms. Modern implementations of TCP contain four intertwined
algorithms: slow start, congestion avoidance, fast retransmit, and fast recovery. In addition to the standard
algorithms used in common implementations of TCP, this paper also describes some of the more common
proposals developed by researchers over the years. We also study, through extensive simulations, the
performance characteristics of four representative TCP schemes, namely TCP Tahoe, New Reno and Vegas
under the network conditions of bottleneck link capacities for wired network
Connection Establishment & Flow and Congestion ControlAdeel Rasheed
On these slides i describe all the detail about Connection Establishment & Flow and Congestion Control. For more detail visit: https://chauhantricks.blogspot.com/
Computer networks have experienced an explosive growth over the past few years and with
that growth have come severe congestion problems. For example, it is now common to see
internet gateways drop 10% of the incoming packets because of local buffer overflows.
Our investigation of some of these problems has shown that much of the cause lies in
transport protocol implementations (
not
in the protocols themselves): The ‘obvious’ ways
to implement a window-based transport protocol can result in exactly the wrong behavior
in response to network congestion. We give examples of ‘wrong’ behavior and describe
some simple algorithms that can be used to make right things happen. The algorithms are
rooted in the idea of achieving network stability by forcing the transport connection to obey
a ‘packet conservation’ principle. We show how the algorithms derive from this principle
and what effect they have on traffic over congested networks.
In October of ’86, the Internet had the first of what became a series of ‘congestion col-
lapses’. During this period, the data throughput from LBL to UC Berkeley (sites separated
by 400 yards and two IMP hops) dropped from 32 Kbps to 40 bps. We were fascinated by
this sudden factor-of-thousand drop in bandwidth and embarked on an investigation of why
things had gotten so bad. In particular, we wondered if the 4.3
BSD
(Berkeley U
NIX
)
TCP
was mis-behaving or if it could be tuned to work better under abysmal network conditions.
The answer to both of these questions was “yes”.
This presentation is created upon mobile sniffers. These mobile sniffers can be used to detect lost mobile phones
This presentation gives the details about the procedure of sniffers functionality and the tools used in producing the device
For the detection of lost mobile SNIFFER plays a vital role.The sniffer device has to be designed precisely and size should be reduced for easy mobility for the purpose of detection .
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.
Comparison of TCP congestion control mechanisms Tahoe, Newreno and VegasIOSR Journals
Abstract: The widely used reliable transport protocol TCP, is an end to end protocol designed for the wireline networks characterized by negligible random packet losses. This paper represents exploratory study of TCP congestion control principles and mechanisms. Modern implementations of TCP contain four intertwined algorithms: slow start, congestion avoidance, fast retransmit, and fast recovery. In addition to the standard algorithms used in common implementations of TCP, this paper also describes some of the more common proposals developed by researchers over the years. We also study, through extensive simulations, the performance characteristics of four representative TCP schemes, namely TCP Tahoe, New Reno and Vegas under the network conditions of bottleneck link capacities for wired network. Keywords - Congestion avoidance, Congestion control mechanisms, Newreno, Tahoe, TCP, Vegas.
1. Distributed Video Streaming Over Internet Thinh PQ Nguyen and Avideh Zakhor Berkeley, CA, USA Presented By Sam
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20. Simulation 20 γ 500bytes Packet size 100ms Ф 50ms Delay for sender 2 0.1*S(i) w 40ms Delay for sender 1
21. Simulations T = 0 to 2s. Starts sending control packets T = 25s, senders send using the algorithms T = 200s, 25 of 100 TCP sources from 1 stop sending T = 400s, 10 new TCP sources start and stop at random
Having multiple senders is also a diversification scheme in that it combats unpredictability of congestion in the Internet.
If the route between a particular sender and the receiver experiences congestion during streaming, the receiver can redistribute streaming rates among other senders, thus resulting in smooth video delivery.
That’s nothing to do with the multi-server video streaming
If there is congestion on a shared link between two senders, then changing the sending rate of one sender may affect the traffic conditions of the other one.
it is preferable to download the entire video in a non-real time fashion before playing it back. The focus is on the the limiting factor in streaming is packet loss and delay due to congestion along the streaming path, rather than the physical bandwidth limitations of the last hop.
the receiver coordinates transmissions from multiple senders based on the information received from the senders.
F(t) total number of loss packet
each sender i also satisfies the constraints in order to share bandwidth fairly with other TCP traffic. The algorithm minimizes F(t) , the number of lost packets during the interval ) , ( . + t t , given instantaneous feedback, and assuming that the estimated loss rate and TCP-friendly available bandwidth are accurate. The proof for this is included in the Appendix.
This is because it takes D(j) for the control packet to arrive ar the sender j N for the kth packet to be sent by sender j D for it arrive at the reeiver sigma
All the senders receive the same control packet from the receive Only use the information in the control packet to update the Ak All use the same equation to do so…r
To illustrate the algorithm, we show a simple example in which, there are two senders, each sending packets at equal rate. As shown in Figure 6, the Sync sequence number is 10. The top line with vertical bars denotes the playback time for the packets. The middle and the bottom lines indicate the time to send packets for senders 1 and 2, respectively. In this scenario, packet 10 will be sent by sender 1 since the estimated difference between playback and its receive time for packet 10 is greater than that of sender 2. Next, packet 11 will be sent by sender 2 for the same reason.
If n is as large as 0.1 then probability is 10-6
Cross/ circle.. Set11… So set 6
the loss rate of scenario two can be 2.5 times larger than that of scenario two. And always over 1 Next, we compare the mean squared error (MSE) of pixel values between the sent frame and the received frame. Since there quite a large number of frames, we further average the MSE for every video frame over a period of 5s. Higher MSE represents lower fidelity of the video due to lost packets. As shown in Figure 14, most of the time the MSE for scenario one lower than that of scenario two.
Finally, there has been work on detecting the shared congestion points of different routes [9] based on the correlation of packet loss and delay between routes. These correlations can be used ahead of time to improve the performance of our approach.