Tcp periodic modeling
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This document discusses mathematical modeling of TCP and factors that affect TCP throughput. It provides an overview of TCP modeling essentials, including modeling window dynamics using linear increase and multiplicative decrease, and modeling packet loss as a stochastic process. It also summarizes some common TCP models, such as the periodic model and detailed packet loss model.
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Tcp vs udp
TCP and UDP are the main Internet protocols for transporting data.
TCP is connection-oriented, reliable, and ensures packets are delivered in order. UDP is simpler and connectionless, meaning it does not ensure delivery or order of packets.
While TCP is slower but reliable, UDP is faster but does not guarantee delivery of packets.
TCP AND UDP
TCP and UDP are transport layer protocols that work in the OSI and TCP/IP models. TCP provides reliable, ordered delivery of data through connection-oriented transmissions, while UDP provides fast but unreliable connectionless delivery. TCP is used for applications like HTTP, FTP, and SMTP that require reliable data transmission. UDP is used for applications like DNS, DHCP, VOIP and online games where losing some data is acceptable but minimizing delay is important. While TCP guarantees delivery, it introduces more overhead; UDP is faster but does not guarantee delivery or order of packets.
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1) ATM uses fixed-sized packets called cells to transfer data at high speeds over both logical connections and physical interfaces in a streamlined manner.
2) ATM cells have a standard size and format including a header for routing and a payload for user data. Multiple logical connections can be multiplexed over a single physical interface.
3) ATM connections can be either virtual channel connections between two end users or virtual path connections that bundle multiple VCCs with the same endpoints to simplify networking and improve performance.
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This document summarizes different algorithms for adaptive retransmission in TCP. The original TCP specification uses an average of sample round-trip times (RTTs) to estimate the RTT and sets the timeout to twice the estimated RTT. The Karn/Partridge algorithm only measures RTT for original transmissions and uses exponential backoff for timeouts after retransmissions. The Jacobson/Karela algorithm calculates estimated RTT and deviation from it to set the timeout to the estimated RTT plus a factor times the deviation, allowing larger timeouts when variance is high.
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The document discusses flow control in TCP. It explains that TCP uses a sliding window mechanism for flow control to balance the sender's transmission rate with the receiver's reception rate. The sliding window allows packets within the window to be transmitted, and slides to the right when acknowledgments are received, making room for more packets. Problems like delayed acknowledgments, silly window syndrome, and solutions like Nagle's algorithm are also covered. TCP provides reliable data transfer using error control mechanisms like checksums, acknowledgments, and retransmissions of lost packets.
Lect9
- TCP uses congestion control and avoidance to prevent network congestion collapse. It operates in a distributed manner without centralized control.
- TCP's congestion control is based on additive increase, multiplicative decrease (AIMD) and uses a congestion window and packet pacing to smoothly increase and decrease transmission rates in response to packet loss as a signal of congestion.
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TCP and UDP are the main Internet protocols for transporting data.
TCP is connection-oriented, reliable, and ensures packets are delivered in order. UDP is simpler and connectionless, meaning it does not ensure delivery or order of packets.
While TCP is slower but reliable, UDP is faster but does not guarantee delivery of packets.
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TCP and UDP are transport layer protocols that work in the OSI and TCP/IP models. TCP provides reliable, ordered delivery of data through connection-oriented transmissions, while UDP provides fast but unreliable connectionless delivery. TCP is used for applications like HTTP, FTP, and SMTP that require reliable data transmission. UDP is used for applications like DNS, DHCP, VOIP and online games where losing some data is acceptable but minimizing delay is important. While TCP guarantees delivery, it introduces more overhead; UDP is faster but does not guarantee delivery or order of packets.
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This document summarizes different algorithms for adaptive retransmission in TCP. The original TCP specification uses an average of sample round-trip times (RTTs) to estimate the RTT and sets the timeout to twice the estimated RTT. The Karn/Partridge algorithm only measures RTT for original transmissions and uses exponential backoff for timeouts after retransmissions. The Jacobson/Karela algorithm calculates estimated RTT and deviation from it to set the timeout to the estimated RTT plus a factor times the deviation, allowing larger timeouts when variance is high.
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The document describes a tool called TCP Congestion Avoidance Algorithm Identification (CAAI) that was proposed to identify the TCP congestion avoidance algorithm of remote web servers. CAAI works in three steps: 1) it gathers TCP window size traces from web servers in emulated network environments, 2) it extracts features like the multiplicative decrease parameter and window growth function from the traces, and 3) it uses these features to classify the TCP algorithm. Testing CAAI on over 30,000 web servers, it was able to identify the default algorithms used by major operating systems, like RENO for Windows and BIC/CUBIC for Linux, as well as some non-default algorithms.
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The document discusses flow control in TCP. It explains that TCP uses a sliding window mechanism for flow control to balance the sender's transmission rate with the receiver's reception rate. The sliding window allows packets within the window to be transmitted, and slides to the right when acknowledgments are received, making room for more packets. Problems like delayed acknowledgments, silly window syndrome, and solutions like Nagle's algorithm are also covered. TCP provides reliable data transfer using error control mechanisms like checksums, acknowledgments, and retransmissions of lost packets.
Lect9
- TCP uses congestion control and avoidance to prevent network congestion collapse. It operates in a distributed manner without centralized control.
- TCP's congestion control is based on additive increase, multiplicative decrease (AIMD) and uses a congestion window and packet pacing to smoothly increase and decrease transmission rates in response to packet loss as a signal of congestion.
- The key mechanisms are slow start for initial rapid ramp up, congestion avoidance for gradual increase, fast retransmit for quick recovery from single losses, and timeout for recovery from multiple losses or ack losses. These mechanisms work together to keep TCP stable and efficient under different network conditions.
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TCP uses congestion control algorithms to dynamically adjust the transmission rate depending on network conditions. It uses three main algorithms:
1. Slow start exponentially increases the congestion window when no congestion is detected.
2. Congestion avoidance additively increases the window when congestion is detected to slow growth.
3. Fast recovery allows additive increases when duplicate ACKs are received, indicating a lost packet but not severe congestion.
TCP detects congestion through timeouts or duplicate ACKs and multiplicatively decreases the window size by half in response to avoid worsening congestion. It transitions between these algorithms depending on congestion signs to maximize throughput while avoiding network overload.
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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.
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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.
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TCP uses congestion control algorithms to dynamically adjust the transmission rate depending on network conditions. It uses three main algorithms:
1. Slow start exponentially increases the congestion window when no congestion is detected.
2. Congestion avoidance additively increases the window when congestion is detected to slow growth.
3. Fast recovery allows additive increases when duplicate ACKs are received, indicating a lost packet but not severe congestion.
TCP detects congestion through timeouts or duplicate ACKs and multiplicatively decreases the window size by half in response to avoid worsening congestion. It transitions between these algorithms depending on congestion signs to maximize throughput while avoiding network overload.
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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”.
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This document summarizes research on congestion control protocols for high-speed networks. It discusses how existing protocols like CUBIC consider every packet drop as congestion and reduce throughput. The document proposes a packet drop guesser module using k-NN to differentiate between packet drops due to congestion versus other factors like noise. It evaluates CUBIC integrated with this module and finds significant performance improvements over CUBIC alone in noisy conditions. Related work on high-speed protocols like BIC, FAST and CUBIC is also summarized.
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¤ TCP Westwood is a congestion control algorithm that improves TCP performance over wireless networks by estimating available bandwidth (BWE) through monitoring ACK packets.
¤ It uses a low-pass filter to average bandwidth measurements and obtain the low-frequency components of available bandwidth. When inter-arrival times of ACKs increase, more weight is given to the most recent bandwidth calculations.
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Watch this Avi webinar to #ByeByeTCPdump forever and learn:
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Watch the full webinar: https://info.avinetworks.com/webinars-avi-tech-corner-episode-1
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PAUSE frames are Ethernet MAC control frames that allow devices on a full-duplex link to request a pause in frame transmission. They carry a PAUSE opcode and pause time parameter, specifying the length of the requested pause in units of 512 bit times. PAUSE frames are sent to a reserved multicast address to simplify flow control without requiring knowledge of the destination's individual address. Switches will not forward frames sent to this multicast address but will process them internally for flow control purposes.
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The document discusses different memory management strategies:
- Swapping allows processes to be swapped temporarily out of memory to disk, then back into memory for continued execution. This improves memory utilization but incurs long swap times.
- Contiguous memory allocation allocates processes into contiguous regions of physical memory using techniques like memory mapping and dynamic storage allocation with first-fit or best-fit. This can cause external and internal fragmentation over time.
- Paging permits the physical memory used by a process to be noncontiguous by dividing memory into pages and mapping virtual addresses to physical frames, allowing more efficient use of memory but requiring page tables for translation.
Tcp Congestion Avoidance
The document discusses several algorithms used for congestion control in TCP/IP networks, including slow start, congestion avoidance, fast retransmit, fast recovery, random early discard (RED), and traffic shaping using leaky bucket and token bucket algorithms. Slow start and congestion avoidance control the transmission rate by adjusting the congestion window size. Fast retransmit and fast recovery allow quicker retransmission of lost packets without waiting for timeouts. RED proactively discards packets before buffer overflow. Leaky bucket and token bucket algorithms shape traffic flow through use of buffers and tokens to smooth bursts and control transmission rates.
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A Packet Drop Guesser Module for Congestion Control Protocols for High speed ...
Different high speed Transport layer protocols have been designed and proposed in the literature to
improve the performance of standard TCP on high BDP links. They are mainly different in their increase
and decrease formulas of their respective congestion control algorithm. Most of these high speed protocols
consider every packet drop in the network as an indication of congestion and they immediately reduce their
congestion window size. Such an approach will usually result in under utilization of available bandwidth in
case of noisy channel conditions. We take CUBIC as a test case and have compared its performance in
case of normal and noisy channel conditions. The throughput of CUBIC was drastically degraded from
50Mbps to 0.5Mbps when we introduced a random packet drops with 0.001 probability. When the
probability of the packet drops increases then the throughput gets decreases. Indeed, we need to
complement existing congestion control algorithms with some intelligent mechanisms that can differentiate
whether a certain packet drop is because of congestion or channel error thus avoid unnecessary window
reduction. In order to distinguish between packets drops, we have developed a k-NN based module to guess
whether the packet drops are due to the congestion or any other reasons. After integrating this module with
CUBIC algorithm, we have observed significant performance improvement.
An Empirical Evaluation of TCP Performance in Online Games
A fundamental design question to ask in the development of a network game is—Which transport protocol should be used—TCP, UDP, or some other protocols? Seeking an objective answer to the choice of communication protocol for MMORPGs, we assess whether TCP, a popular choice, is suitable for MMORPGs based on empirical evidence. To the best of our knowledge, this work is the first evaluation of transport protocol performance using real-life game traces.
We analyze a 1, 356-million-packet trace from ShenZhou Online, a TCP-based, commercial, mid-sized MMORPG. Our analysis indicates that TCP is unwieldy and inappropriate for MMORPGs. This is due to four distinctive characteristics of MMORPG traffic: 1) tiny packets, 2) low packet rate, 3) application-limited traffic generation, and 4) bidirectional traffic. We show that because TCP was originally designed for unidirectional and network-limited bulk data transfers, it cannot adapt well to MMORPG traffic. In particular, the window-based congestion control and the fast retransmit algorithm for loss recovery are ineffective. Furthermore, TCP is overkill, as not every game packet needs to be transmitted in a reliably and orderly manner. We also show that the degraded network performance did impact users’ willingness to continue a game. Finally, we discuss guidelines in designing transport protocols for online games.
TCP Theory
TCP provides reliable data transfer over unreliable packet networks by using acknowledgments, retransmissions, and adaptive congestion control. It works with IP to transfer data through routers that may drop packets. While TCP ensures reliable delivery, it must control its transmission rate to avoid overwhelming network capacity and causing congestion collapse. This is achieved through additive-increase, multiplicative-decrease of the congestion window and techniques like active queue management.
chapter 3.2 TCP.pptx
TCP provides reliable data transfer through several key features:
- It numbers data bytes and uses acknowledgments to ensure all bytes are received correctly. If bytes are lost, they are retransmitted.
- Congestion control algorithms like slow start and congestion avoidance allow TCP to gradually increase data transfer rates while avoiding overwhelming the network.
- Fast retransmit detects lost packets sooner by retransmitting on three duplicate ACKs, while fast recovery resumes data transfer using ACKs still in the pipe.
Lecture 19 22. transport protocol for ad-hoc
This document discusses transport layer protocols for mobile ad hoc networks (MANETs). It begins with an introduction to MANETs and the need for new network architectures and protocols to support new types of networks. It then provides an overview of TCP/IP and how TCP works, including congestion control mechanisms. The document discusses challenges for TCP over wireless networks, where packet losses are often due to errors rather than congestion. It covers different versions of TCP and their approaches to congestion control. The goal is to design transport layer protocols that can address the unreliable links and frequent topology changes in MANETs.
04 MK-PPT End-to-End Protocols.ppt
This chapter discusses transport layer protocols and how they enable process-to-process communication across a network. It introduces the two main transport protocols: UDP, which provides simple demultiplexing of data without reliability, and TCP, which provides reliable, connection-oriented data transmission. The chapter outlines how TCP establishes connections, provides flow and congestion control using sliding window algorithms, and ensures reliable data delivery despite properties of the underlying network that can result in dropped, reordered, or duplicated packets.
Iaetsd an effective approach to eliminate tcp incast
This document proposes an Incast Congestion Control for TCP (ICTCP) scheme to eliminate TCP incast collapse in datacenter environments. TCP incast collapse occurs when multiple synchronized servers send data to the same receiver in parallel, overwhelming the switch buffer and causing packet loss. ICTCP is a receiver-side approach that proactively adjusts the TCP receive window size of connections to control their aggregate burstiness and prevent switch buffer overflow before packet loss occurs. It estimates available bandwidth and uses this as a quota to coordinate receive window increases. For each connection, the receive window is adjusted based on the ratio of the difference between measured and expected throughput. This allows adaptive tuning of receive windows to meet sender throughput needs while avoiding congest
Computer network (13)
This document summarizes key concepts about congestion control in TCP including:
- TCP uses additive increase multiplicative decrease (AIMD) to dynamically adjust the congestion window size and maintain efficiency and fairness.
- TCP has slow start and congestion avoidance states that govern how the congestion window is adjusted in response to acknowledgements.
- TCP responds to packet loss through fast retransmit, fast recovery, and halving the congestion window size to reduce congestion according to protocols like Tahoe, Reno, and New Reno.
UDT
The document summarizes the UDT protocol, which is a high performance transport protocol designed for data-intensive applications over high-speed networks. It discusses the limitations of TCP for these applications and high bandwidth-delay product networks. It then provides an overview of the design and implementation of the UDT protocol, including its congestion control algorithm, APIs, and composable framework. It evaluates UDT's performance in terms of efficiency, fairness, and stability compared to TCP. The goal of UDT is to enable efficient, fair, and friendly transport of data for distributed applications over high-speed networks.
Designing TCP-Friendly Window-based Congestion Control
TFWC is a proposed window-based congestion control algorithm that is designed to be TCP-friendly for real-time multimedia applications, while addressing some issues with the standard rate-based TFRC algorithm. TFWC uses a TCP-like acknowledgment clock and window sizing equation to achieve smooth throughput similar to TFRC, but provides better fairness when competing with TCP traffic and is simpler to implement without needing to measure round-trip times. Analysis shows that TFWC provides fairness comparable to TFRC, smoothness on par with TFRC, and faster responsiveness to changes in available bandwidth.
Improving Performance of TCP in Wireless Environment using TCP-P
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.
UDT
The document describes the design and implementation of a new high performance data transport protocol called UDT. UDT is implemented at the application layer over UDP to provide reliable, high-speed data transfer capabilities. It includes a new congestion control algorithm based on AIMD with decreasing increases that aims for efficiency, fairness and friendliness. Experimental results show UDT achieves high throughput and good fairness compared to TCP. The document also introduces a configurable framework called Composable UDT that allows new congestion control algorithms to be easily implemented and evaluated.
Mobile Transpot Layer
This document discusses various transport layer protocols for mobile networks. It begins by describing TCP and its mechanisms for congestion avoidance, flow control, slow start, and retransmission. It then covers several TCP variants including Tahoe, Reno, and Vegas. It also discusses indirect TCP, Snoop TCP, and Mobile TCP which aim to optimize TCP for wireless networks by handling retransmissions locally or splitting the connection. The document provides details on the algorithms and functioning of these different protocols.
TCP Over Wireless
This document discusses TCP over wireless networks. It explains that TCP was designed for fixed networks with low delay and errors, but wireless networks have high delay, errors and variable bandwidth. This causes TCP to perform poorly over wireless. The document outlines various techniques to improve TCP performance over wireless like Fast Retransmit and Recovery, Slow Start proposals with larger initial windows, ACK counting and ACK-every-segment. It also discusses protocols like HTTP, RLP that operate between TCP and the wireless transmission layers.
Alternative Transport Protocols
Overview of transport protocols as alternatives to TCP and UDP.
TCP and UDP are the two transport protocols (OSI layer 4) that are predominantly used by applications in IP based networks.
The properties of TCP and UDP are complementary in that TCP provides many quality of service features that UDP lacks.
Therefore, TCP is mainly used in applications that require a certain level of reliable transport connection while UDP is used when reliability is of secondary importance but speed and simplicity are important.
There are, however, alternatives to TCP and UDP. SCTP (Stream Control Transmission Protocol) was defined some time ago and was meant to eventually replace TCP. It provides the same features as TCP but fixes some of the shortcomings of TCP. Alternatives for UDP exist as well such as Reliable UDP and UDP redundancy.
Tcp variants for data center networks
It is a rare PPT in slide share , and it will help you a lot to make sense about TCP VARIANTS IN DATA CENTER NETWORK
TLS in manet
The document discusses challenges with using TCP in mobile ad hoc networks (MANETs) and evaluates potential solutions. Specifically, it finds that:
1) TCP performs poorly in MANETs due to high packet loss from route failures and wireless errors, which TCP misinterprets as congestion.
2) TCP variants like Westwood and Jersey that more accurately estimate bandwidth perform better but are not sufficient.
3) A new transport protocol like ATP that is rate-based rather than window-based and leverages intermediate nodes may better address MANET issues.
Jaimin chp-6 - transport layer- 2011 batch
The document discusses the transport layer in computer networking. It explains that the transport layer provides logical communication between processes running on different hosts. It describes two main transport protocols: TCP and UDP. TCP provides connection-oriented transmission that is reliable and in-order, while UDP provides connectionless transmission that is unreliable and unordered. The document also covers topics like connection establishment, port numbers, sockets, and services provided by the transport layer.
Lec 4(packet delay layered Architecture)
This document summarizes key points from a lecture on computer networks:
1) There are various delays that packets experience at each node in a network, including processing delay, queueing delay, transmission delay, and propagation delay.
2) Transmission delay, the time to push all bits through a link, depends on packet length and link bandwidth. Propagation delay, the time for a signal to propagate between nodes, depends on the distance and propagation speed.
3) The traceroute program is used to measure delays from the source to each router along the path to the destination, demonstrating increasing delays along the path.
4) Network layers, such as application, transport, network, and link layers,
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A Packet Drop Guesser Module for Congestion Control Protocols for High speed ...
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An Empirical Evaluation of TCP Performance in Online Games
An Empirical Evaluation of TCP Performance in Online Games
Iaetsd an effective approach to eliminate tcp incast
Iaetsd an effective approach to eliminate tcp incast
Designing TCP-Friendly Window-based Congestion Control
Designing TCP-Friendly Window-based Congestion Control
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Disclaimer: No one is perfect, so please mind that there might be mistakes and typos.
dtubbenhauer@gmail.com
Corrected slides: dtubbenhauer.com/talks.html
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Tcp periodic modeling
- 1. Submitted by: Varsha Anandani (13MIT0062)
- 2. TCP = Transmission Control Protocol Connection-oriented protocol Provides a reliable unicast end-to-end byte stream over an unreliable internetwork
- 3. TCP
- 4. Motivation for mathematical modeling of TCP 1) The sheer scale of systems in which we find TCP operating is tremendous. 2) There still remain many unknowns in the environment in which TCP is operating. 3) TCP was not designed using optimization.
- 5. Motivation for TCP Modeling TCP operating scale is very large Models are required to gain deeper understanding of TCP dynamics Uncertainties can be modeled as stochastic processes Drive the design of TCP-friendly algorithms for multimedia applications Optimize TCP performance
- 6. FACTORS THAT AFFECT THROUGHPUT Link speed the number of bits per second that can be transmitted. Propagation delay, the time it takes the actual electronic, optical, or other signals to travel from one end of the connection to the other. Window size, the amount of unacknowledged data that can be outstanding on a TCP connection. Link reliability. Network and intermediate device congestion. Path maximum transmission unit (PMTU).
- 7. TCP Modeling Essentials Mainly Reno flavors are modeled Two main features are modeled Window dynamics Packet loss process
- 8. Window Dynamics Linear increase and multiplicative decrease is modelled. The standard assumption X(t) = W(t)/RTT
- 9. Packet Loss Process Packet loss triggers window decrease Packet loss is uncertain This uncertainty is typically modeled as a stochastic process E.g. probability p of losing a packet
- 10. Gallery of TCP Models Periodic model Detailed packet loss model Finite state machine Fluid flow model And others…
- 11. Periodic Model Simplest model for TCP No specific version assumed Assumes a periodic pattern of congestion window X(p) = (1/RTT)*Root(3/2p) P is the packet loss probability
- 12. TCP Congestion Control: window algorithm Window: can send W packets increase window by one per RTT if no loss, W <- W+1 each RTT decrease window by half on detection of loss W <- W/2 receiver W sender 1 RTT
- 13. TCP throughput/loss relationship Idealized model: W is maximum supportable window size (then loss occurs) TCP window starts at w/2 grows to W, then halves, then grows to W, then halves… one window worth of packets each rtt to find: throughput as function of loss, RTT W TCP window size time (rtt) W/2 loss occurs period
- 14. period # packets sent per “period” 1 1 3 W W W 2 2 2 8 8 4
- 15. Example 1 - Question If a TCP connection has an average RTT of 200ms, and packet loss probability 0.05, what is the average throughput?