Jose Saldana, Julian Fernandez-Navajas, Jose Ruiz-Mas "Can We Multiplex ACKs without Harming the Performance of TCP?," Consumer Communications and Networking Conference, CCNC 2014. Las Vegas, January 10, 2014, pp 921-922. ISBN 978-1-4799-2356-4.
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.
Optimization of Low-efficiency Traffic in OpenFlowSoftware Defined NetworksJose Saldana
This paper proposes a method for optimizing bandwidth usage in Software Defined Networks (SDNs) based on OpenFlow. Flows of small packets presenting a high overhead, as the ones generated by emerging services, can be identified by the SDN controller, in order to remove header fields that are common to any packet in the flow, only during their way through the SDN. At the same time, several packets can be multiplexed together in the same frame, thus reducing the number of sent frames. Four kinds of small-packet traffic flows are considered (VoIP, UDP and TCP-based online games, and ACKs from TCP flows). Both IPv4 and IPv6 are tested, and significant bandwidth savings (up to 68 % for IPv4 and 78 % for IPv6) can be obtained for the considered kinds of traffic.
Early-stage topological and technological choices for TSN-based communication...RealTime-at-Work (RTaW)
A main issue in the design of automotive communication architectures is that the most important design choices pertaining to the topology of the networks and the technologies to use (protocols, data rate, hardware) have to be made at a time when the communication requirements are not entirely known. Indeed, many functions only becomes available along the development cycle, and vehicle platforms have to support incremental evolutions of the embedded system that may not be fully foreseeable at the time design choices are made. The problem is becoming even more difficult and crucial with the introduction of dynamically evolving communication requirements requiring network re-configuration at run-time.
We present how the use of synthetic data, that is data generated programmatically based on past vehicle projects and what can be foreseen for the current project, enables the designers to make such early stage choices based on quantified metrics. The proposals are applied to Groupe Renault's FACE service-oriented E/E architecture with the use of the “Topology Stress Test” feature implemented in RTaW-Pegase.
This paper proposes a new end-to-end congestion control protocol called ACP that is designed for high bandwidth-delay product networks. ACP aims to achieve high link utilization, fairness among flows, and fast convergence. It does this by estimating the bottleneck queue size upon detecting congestion and decreasing the congestion window by exactly the amount needed to empty the queue. It also uses a "fairness ratio" metric to determine window increases to ensure convergence to a fair share of bandwidth among flows. The paper argues that existing protocols cannot achieve high utilization and fairness due to their inability to accurately measure link load. It claims ACP addresses this through a new congestion window control approach combining queue size estimation and a fairness measure.
The Effect of Multiplexing Delay on MMORPG TCP Traffic FlowsJose Saldana
Jose Saldana, "The Effect of Multiplexing Delay on MMORPG TCP Traffic Flows," Consumer Communications and Networking Conference, CCNC 2014. Las Vegas, January 10, 2014, pp 447-452. ISBN 978-1-4799-2356-4.
Fast channel zapping with destination oriented multicast for ip video deliveryecway
This document discusses a destination-oriented multicast assisted zapping acceleration (DAZA) scheme for IP video delivery. It aims to improve channel zapping time, which is an important quality metric. The DAZA scheme uses time-shifted subchannels to ensure fast zapping within a delay bound while maintaining picture quality. It analyzes the optimal subchannel data rate and addresses a startup effect issue. DAZA implements scalable destination-oriented multicast instead of traditional IP multicast. Simulation results demonstrate that DAZA validates the analysis and improves robustness and messaging overhead in distributed environments.
Influence of Online Games Traffic Multiplexing and Router Buffer on Subjectiv...Jose Saldana
This document discusses the influence of online games, traffic multiplexing, and router buffer on subjective quality. It presents the results of tests on multiplexing gaming traffic using different period sizes and router buffer configurations. The key findings are:
- Multiplexing can reduce bandwidth and packets per second by up to 30% and 35% respectively, at the cost of increased delay and jitter.
- Larger period sizes and router buffers increase delay more than smaller configurations. Multiplexing is limited by the bandwidth of around 1400kbps.
- Jitter also increases with multiplexing but is limited by the bandwidth, while smaller router buffers introduce less jitter and delay.
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.
Optimization of Low-efficiency Traffic in OpenFlowSoftware Defined NetworksJose Saldana
This paper proposes a method for optimizing bandwidth usage in Software Defined Networks (SDNs) based on OpenFlow. Flows of small packets presenting a high overhead, as the ones generated by emerging services, can be identified by the SDN controller, in order to remove header fields that are common to any packet in the flow, only during their way through the SDN. At the same time, several packets can be multiplexed together in the same frame, thus reducing the number of sent frames. Four kinds of small-packet traffic flows are considered (VoIP, UDP and TCP-based online games, and ACKs from TCP flows). Both IPv4 and IPv6 are tested, and significant bandwidth savings (up to 68 % for IPv4 and 78 % for IPv6) can be obtained for the considered kinds of traffic.
Early-stage topological and technological choices for TSN-based communication...RealTime-at-Work (RTaW)
A main issue in the design of automotive communication architectures is that the most important design choices pertaining to the topology of the networks and the technologies to use (protocols, data rate, hardware) have to be made at a time when the communication requirements are not entirely known. Indeed, many functions only becomes available along the development cycle, and vehicle platforms have to support incremental evolutions of the embedded system that may not be fully foreseeable at the time design choices are made. The problem is becoming even more difficult and crucial with the introduction of dynamically evolving communication requirements requiring network re-configuration at run-time.
We present how the use of synthetic data, that is data generated programmatically based on past vehicle projects and what can be foreseen for the current project, enables the designers to make such early stage choices based on quantified metrics. The proposals are applied to Groupe Renault's FACE service-oriented E/E architecture with the use of the “Topology Stress Test” feature implemented in RTaW-Pegase.
This paper proposes a new end-to-end congestion control protocol called ACP that is designed for high bandwidth-delay product networks. ACP aims to achieve high link utilization, fairness among flows, and fast convergence. It does this by estimating the bottleneck queue size upon detecting congestion and decreasing the congestion window by exactly the amount needed to empty the queue. It also uses a "fairness ratio" metric to determine window increases to ensure convergence to a fair share of bandwidth among flows. The paper argues that existing protocols cannot achieve high utilization and fairness due to their inability to accurately measure link load. It claims ACP addresses this through a new congestion window control approach combining queue size estimation and a fairness measure.
The Effect of Multiplexing Delay on MMORPG TCP Traffic FlowsJose Saldana
Jose Saldana, "The Effect of Multiplexing Delay on MMORPG TCP Traffic Flows," Consumer Communications and Networking Conference, CCNC 2014. Las Vegas, January 10, 2014, pp 447-452. ISBN 978-1-4799-2356-4.
Fast channel zapping with destination oriented multicast for ip video deliveryecway
This document discusses a destination-oriented multicast assisted zapping acceleration (DAZA) scheme for IP video delivery. It aims to improve channel zapping time, which is an important quality metric. The DAZA scheme uses time-shifted subchannels to ensure fast zapping within a delay bound while maintaining picture quality. It analyzes the optimal subchannel data rate and addresses a startup effect issue. DAZA implements scalable destination-oriented multicast instead of traditional IP multicast. Simulation results demonstrate that DAZA validates the analysis and improves robustness and messaging overhead in distributed environments.
Influence of Online Games Traffic Multiplexing and Router Buffer on Subjectiv...Jose Saldana
This document discusses the influence of online games, traffic multiplexing, and router buffer on subjective quality. It presents the results of tests on multiplexing gaming traffic using different period sizes and router buffer configurations. The key findings are:
- Multiplexing can reduce bandwidth and packets per second by up to 30% and 35% respectively, at the cost of increased delay and jitter.
- Larger period sizes and router buffers increase delay more than smaller configurations. Multiplexing is limited by the bandwidth of around 1400kbps.
- Jitter also increases with multiplexing but is limited by the bandwidth, while smaller router buffers introduce less jitter and delay.
The document summarizes performance tests comparing the eXtreme TCP (XTCP) protocol to standard TCP and other TCP variants. Automated tests transferred a 64MB file between servers located around the world via FTP. XTCP consistently achieved download rates 5-13 times faster than standard TCP and showed small performance gains over TCP variants like Vegas, Cubic, and HTCP in most test scenarios. XTCP was able to better detect and utilize available network bandwidth, especially over high latency connections.
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.
This document provides an overview of TCP congestion control algorithms. It describes the basic additive increase/multiplicative decrease approach and key mechanisms like slow start, fast retransmit, and fast recovery. It also discusses algorithms for setting the retransmission timeout value and adaptations made in protocols like New Reno and Cubic.
1) Scheduling CAN frames with offsets provides a major performance boost by desynchronizing transmissions to avoid load peaks and make CAN networks more predictable at higher loads over 60%.
2) Computing the optimal offsets is an exponential problem but approximate algorithms can efficiently assign offsets to reduce worst-case response times, often cutting them by over half.
3) The NETCAR-Analyzer software can compute frame response times on CAN networks with offsets and help dimension system buffers using a fast implementation.
- 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.
Influence of the Distribution of TCRTP Multiplexed Flows on VoIP Conversation...Jose Saldana
This document describes an experiment to evaluate the impact of different Tunneling Compressed RTP (TCRTP) multiplexing schemes on voice quality over IP (VoIP) under varying network conditions. The experiment multiplexed VoIP packets using TCRTP tunnels with different numbers of flows and measured the resulting voice quality using the R-factor metric. With a high-capacity router buffer, all TCRTP schemes showed step-like quality degradation as background traffic increased. With a time-limited buffer, smaller tunnels led to smoother quality decline. More flows per tunnel reduced overhead and allowed higher background traffic levels before quality dropped.
The project focuses on how Congestion Control and Queue Management techniques have evolved in the course of time and being modified to minimize packet loss and stabilize Queue length
Improving Distributed TCP Caching for Wireless Sensor NetworksAhmed Ayadi
The document proposes an enhanced distributed TCP caching (EDTC) approach to improve TCP performance over wireless sensor networks. EDTC improves upon distributed TCP caching (DTC) by detecting and handling TCP acknowledgment losses, disabling unnecessary retransmissions, and using a smoothed retransmission timeout value. Simulation results show that EDTC reduces energy consumption and transfer duration compared to DTC and TCP, especially in high packet loss networks.
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.
Analytical Research of TCP Variants in Terms of Maximum ThroughputIJLT EMAS
This paper is comparative, throughput analysis, for
the TCP variants as for New Reno, Westwood & High Speed,
and it analyzes the outcomes in simulated environment for NS -3
(version 3.25) simulator with reference to multiple varying
network parameters that includes network simulation time,
router bandwidth, varying traffic source counts to observe which
is one of the best TCP variant in different scenarios. Analysis
was done using dumbbell topology to figure out the comparative
maximum throughput of TCP variants. The analysis gives result
as TCP Variant “NewReno” is good when low bandwidth is used,
while TCP Variant “HighS peed” is good in terms of using large
bandwidths in comparison to Westwood. Network traffic flow
was observed in NetAnim tool.
Delay jitter control for real time communicationMasud Rana
This document proposes a method for controlling delay jitter for real-time communication channels in a packet-switching network. It extends an existing scheme that provides bounds on maximum delay. The key aspects are:
1) Each network node contains "regulators" for each channel that reconstruct the original packet arrival pattern to preserve jitter, and a scheduler that ensures low distortion of patterns between nodes.
2) Clients specify maximum delay and jitter bounds when establishing a channel. The establishment procedure sets local jitter bounds at each node to ensure the end-to-end bound is met.
3) Regulators at each node attempt to faithfully preserve the original packet arrival pattern, so that the last node sees essentially the original
"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.
TCP uses congestion control to determine how much capacity is available in the network and regulate how many packets can be in transit. It uses additive increase/multiplicative decrease (AIMD) where the congestion window is increased slowly with each ACK but halved upon timeout. Slow start is used initially and after idle periods to grow the window exponentially until congestion is detected. Fast retransmit and fast recovery help detect and recover from packet loss without requiring a timeout.
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.
This document presents an overview of computer network congestion and congestion control techniques. It defines congestion as occurring when too many packets are present in a network link, causing queues to overflow and packets to drop. It then discusses factors that can cause congestion as well as the costs. It outlines open-loop and closed-loop congestion control approaches. Specific algorithms covered include leaky bucket, token bucket, choke packets, hop-by-hop choke packets, and load shedding. The document concludes by noting the importance of efficient congestion control techniques with room for improvement.
The document discusses several ways to optimize TCP/IP network performance for high-bandwidth connections. It recommends using large MTUs, tuning TCP window sizes based on bandwidth-delay products, enabling features like SACK and window scaling, and using queue management techniques like RED to reduce packet loss. Proper configuration of these TCP parameters is important for achieving high throughput over high-speed networks.
TCP Performance analysis Wireless Multihop NetworksAbhishek Kona
This document summarizes an experiment analyzing TCP performance over multi-hop wireless networks using a test bed. The experiment varied hop count, window size, and TCP variants. Results showed degradation in throughput with increased hops. Throughput peaked at certain window sizes depending on hops. WESTWOOD marginally outperformed other variants with small windows. Turning off TCP SACK gave better performance over 3 hops. Limitations included node availability and high data variance. Previous studies are difficult to emulate fully in real deployments.
performance evaluation of TCP varients in Mobile ad-hoc Networkခ်စ္ စု
This document summarizes a presentation given at the Ninth National Conference on Science and Engineering in Upper Myanmar on analyzing the performance of TCP variants (Tahoe, Reno, Vegas) in mobile ad hoc networks (MANETs) under different scenarios. The presentation aimed to identify the most suitable TCP variant for MANETs. It discussed TCP variants, the network and simulation model, scenarios varying node count and speed, and results showing that TCP Vegas generally performed best in terms of throughput.
The Utility of Characterizing Packet Loss as a Function of Packet Size in Com...Jose Saldana
Jose Saldana, Julian Fernandez-Navajas, Jose Ruiz-Mas, Eduardo Viruete Navarro, Luis Casadesus, "The Utility of Characterizing Packet Loss as a Function of Packet Size in Commercial Routers," in Proc. CCNC 2012, Work in progress papers, pp. 362-363, Las Vegas. Jan 2012. ISBN 9781457720697
Improving Network Efficiency with SimplemuxJose Saldana
Jose Saldana, Ignacio Forcen, Julian Fernandez-Navajas, Jose Ruiz-Mas, "Improving Network Efficiency with Simplemux,'' IEEE CIT 2015, International Conference on Computer and Information Technology, 26-28 October 2015 in Liverpool, UK. (http://cse.stfx.ca/~cit2015/)
Presentation the paper http://diec.unizar.es/~jsaldana/personal/chicago_CIT2015_in_proc.pdf
Abstract
The high amount of small packets currently transported by IP networks results in a high overhead, caused by the significant header-to-payload ratio of these packets. In addition, the MAC layer of wireless technologies makes a non-optimal use of airtime when packets are small. Small packets are also costly in terms of processing capacity. This paper presents Simplemux, a protocol able to multiplex a number of packets sharing a common network path, thus increasing efficiency when small packets are transported. It can be useful in constrained scenarios where resources are scarce, as community wireless networks or IoT. Simplemux can be seen as an alternative to Layer-2 optimization, already available in 802.11 networks. The design of Simplemux is presented, and its efficiency improvement is analyzed. An implementation is used to carry out some tests with real traffic, showing significant improvements: 46% of the bandwidth can be saved when compressing voice traffic; the reduction in terms of packets per second in an Internet trace can be up to 50%. In wireless networks, packet grouping results in a significantly improved use of air time.
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.
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.
The document summarizes performance tests comparing the eXtreme TCP (XTCP) protocol to standard TCP and other TCP variants. Automated tests transferred a 64MB file between servers located around the world via FTP. XTCP consistently achieved download rates 5-13 times faster than standard TCP and showed small performance gains over TCP variants like Vegas, Cubic, and HTCP in most test scenarios. XTCP was able to better detect and utilize available network bandwidth, especially over high latency connections.
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.
This document provides an overview of TCP congestion control algorithms. It describes the basic additive increase/multiplicative decrease approach and key mechanisms like slow start, fast retransmit, and fast recovery. It also discusses algorithms for setting the retransmission timeout value and adaptations made in protocols like New Reno and Cubic.
1) Scheduling CAN frames with offsets provides a major performance boost by desynchronizing transmissions to avoid load peaks and make CAN networks more predictable at higher loads over 60%.
2) Computing the optimal offsets is an exponential problem but approximate algorithms can efficiently assign offsets to reduce worst-case response times, often cutting them by over half.
3) The NETCAR-Analyzer software can compute frame response times on CAN networks with offsets and help dimension system buffers using a fast implementation.
- 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.
Influence of the Distribution of TCRTP Multiplexed Flows on VoIP Conversation...Jose Saldana
This document describes an experiment to evaluate the impact of different Tunneling Compressed RTP (TCRTP) multiplexing schemes on voice quality over IP (VoIP) under varying network conditions. The experiment multiplexed VoIP packets using TCRTP tunnels with different numbers of flows and measured the resulting voice quality using the R-factor metric. With a high-capacity router buffer, all TCRTP schemes showed step-like quality degradation as background traffic increased. With a time-limited buffer, smaller tunnels led to smoother quality decline. More flows per tunnel reduced overhead and allowed higher background traffic levels before quality dropped.
The project focuses on how Congestion Control and Queue Management techniques have evolved in the course of time and being modified to minimize packet loss and stabilize Queue length
Improving Distributed TCP Caching for Wireless Sensor NetworksAhmed Ayadi
The document proposes an enhanced distributed TCP caching (EDTC) approach to improve TCP performance over wireless sensor networks. EDTC improves upon distributed TCP caching (DTC) by detecting and handling TCP acknowledgment losses, disabling unnecessary retransmissions, and using a smoothed retransmission timeout value. Simulation results show that EDTC reduces energy consumption and transfer duration compared to DTC and TCP, especially in high packet loss networks.
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.
Analytical Research of TCP Variants in Terms of Maximum ThroughputIJLT EMAS
This paper is comparative, throughput analysis, for
the TCP variants as for New Reno, Westwood & High Speed,
and it analyzes the outcomes in simulated environment for NS -3
(version 3.25) simulator with reference to multiple varying
network parameters that includes network simulation time,
router bandwidth, varying traffic source counts to observe which
is one of the best TCP variant in different scenarios. Analysis
was done using dumbbell topology to figure out the comparative
maximum throughput of TCP variants. The analysis gives result
as TCP Variant “NewReno” is good when low bandwidth is used,
while TCP Variant “HighS peed” is good in terms of using large
bandwidths in comparison to Westwood. Network traffic flow
was observed in NetAnim tool.
Delay jitter control for real time communicationMasud Rana
This document proposes a method for controlling delay jitter for real-time communication channels in a packet-switching network. It extends an existing scheme that provides bounds on maximum delay. The key aspects are:
1) Each network node contains "regulators" for each channel that reconstruct the original packet arrival pattern to preserve jitter, and a scheduler that ensures low distortion of patterns between nodes.
2) Clients specify maximum delay and jitter bounds when establishing a channel. The establishment procedure sets local jitter bounds at each node to ensure the end-to-end bound is met.
3) Regulators at each node attempt to faithfully preserve the original packet arrival pattern, so that the last node sees essentially the original
"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.
TCP uses congestion control to determine how much capacity is available in the network and regulate how many packets can be in transit. It uses additive increase/multiplicative decrease (AIMD) where the congestion window is increased slowly with each ACK but halved upon timeout. Slow start is used initially and after idle periods to grow the window exponentially until congestion is detected. Fast retransmit and fast recovery help detect and recover from packet loss without requiring a timeout.
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.
This document presents an overview of computer network congestion and congestion control techniques. It defines congestion as occurring when too many packets are present in a network link, causing queues to overflow and packets to drop. It then discusses factors that can cause congestion as well as the costs. It outlines open-loop and closed-loop congestion control approaches. Specific algorithms covered include leaky bucket, token bucket, choke packets, hop-by-hop choke packets, and load shedding. The document concludes by noting the importance of efficient congestion control techniques with room for improvement.
The document discusses several ways to optimize TCP/IP network performance for high-bandwidth connections. It recommends using large MTUs, tuning TCP window sizes based on bandwidth-delay products, enabling features like SACK and window scaling, and using queue management techniques like RED to reduce packet loss. Proper configuration of these TCP parameters is important for achieving high throughput over high-speed networks.
TCP Performance analysis Wireless Multihop NetworksAbhishek Kona
This document summarizes an experiment analyzing TCP performance over multi-hop wireless networks using a test bed. The experiment varied hop count, window size, and TCP variants. Results showed degradation in throughput with increased hops. Throughput peaked at certain window sizes depending on hops. WESTWOOD marginally outperformed other variants with small windows. Turning off TCP SACK gave better performance over 3 hops. Limitations included node availability and high data variance. Previous studies are difficult to emulate fully in real deployments.
performance evaluation of TCP varients in Mobile ad-hoc Networkခ်စ္ စု
This document summarizes a presentation given at the Ninth National Conference on Science and Engineering in Upper Myanmar on analyzing the performance of TCP variants (Tahoe, Reno, Vegas) in mobile ad hoc networks (MANETs) under different scenarios. The presentation aimed to identify the most suitable TCP variant for MANETs. It discussed TCP variants, the network and simulation model, scenarios varying node count and speed, and results showing that TCP Vegas generally performed best in terms of throughput.
The Utility of Characterizing Packet Loss as a Function of Packet Size in Com...Jose Saldana
Jose Saldana, Julian Fernandez-Navajas, Jose Ruiz-Mas, Eduardo Viruete Navarro, Luis Casadesus, "The Utility of Characterizing Packet Loss as a Function of Packet Size in Commercial Routers," in Proc. CCNC 2012, Work in progress papers, pp. 362-363, Las Vegas. Jan 2012. ISBN 9781457720697
Improving Network Efficiency with SimplemuxJose Saldana
Jose Saldana, Ignacio Forcen, Julian Fernandez-Navajas, Jose Ruiz-Mas, "Improving Network Efficiency with Simplemux,'' IEEE CIT 2015, International Conference on Computer and Information Technology, 26-28 October 2015 in Liverpool, UK. (http://cse.stfx.ca/~cit2015/)
Presentation the paper http://diec.unizar.es/~jsaldana/personal/chicago_CIT2015_in_proc.pdf
Abstract
The high amount of small packets currently transported by IP networks results in a high overhead, caused by the significant header-to-payload ratio of these packets. In addition, the MAC layer of wireless technologies makes a non-optimal use of airtime when packets are small. Small packets are also costly in terms of processing capacity. This paper presents Simplemux, a protocol able to multiplex a number of packets sharing a common network path, thus increasing efficiency when small packets are transported. It can be useful in constrained scenarios where resources are scarce, as community wireless networks or IoT. Simplemux can be seen as an alternative to Layer-2 optimization, already available in 802.11 networks. The design of Simplemux is presented, and its efficiency improvement is analyzed. An implementation is used to carry out some tests with real traffic, showing significant improvements: 46% of the bandwidth can be saved when compressing voice traffic; the reduction in terms of packets per second in an Internet trace can be up to 50%. In wireless networks, packet grouping results in a significantly improved use of air time.
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.
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.
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 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.
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Implementation and Performance Analysis of a UDP Binding for SOAPDr. Fahad Aijaz
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performance as of TCP-Reno and TCP-New Reno. But we can
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environment.
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Can We Multiplex ACKs without Harming the Performance of TCP?
1. CCNC 2014, The 11Th Annual IEEE Consumer Communications & Networking Conference
January 10-13 Las Vegas, Nevada USA
Can We Multiplex ACKs
without Harming the
Performance of TCP?
Jose Saldana, Julián Fernández-Navajas, José Ruiz-Mas
2. Index
1. Introduction
2. Tests and results
3. Conclusions
Can We Multiplex ACKs without Harming the Performance of TCP? - CCNC 2014
3. Index
1. Introduction
2. Tests and results
3. Conclusions
Can We Multiplex ACKs without Harming the Performance of TCP? - CCNC 2014
4. Introduction
Increase of emerging real-time services
They use small packets
This is modifying the traffic mix present
on the Internet
Inefficiency of the packets
IPv6 makes the problem even worse
VoIP packets
Can We Multiplex ACKs without Harming the Performance of TCP? - CCNC 2014
5. Introduction
TCRTP (RFC4170) improves the efficiency
of VoIP. It uses three layers:
Header compression
Multiplexing
Tunneling
IP network
.
.
.
MUX
RTP
DEMUX
RTP multiplexing
Can We Multiplex ACKs without Harming the Performance of TCP? - CCNC 2014
RTP
.
.
.
6. Introduction
Advantage: Bandwidth and pps savings
At the cost of an additional multiplexing delay
Native VoIP
traffic
...
Inter-pkt time
Inter- pkt time
Inter-pkt time
...
...
...
...
...
Optimized
traffic
...
Can We Multiplex ACKs without Harming the Performance of TCP? - CCNC 2014
...
7. Introduction
TCM-TF*: Proposal for multiplexing other traffic
flows, including UDP (non-RTP) and TCP
payload
payload
RTP
payload
payload
RTP
UDP
TCP
UDP
UDP
IP
IP
IP
IP
ECRTP
Compression layer
No compr. / ROHC / IPHC / ECRTP
PPPMux
Multiplexing layer
PPPMux / Other
L2TP
Tunneling layer
GRE / L2TP
MPLS
IP
a) TCRTP
Network Protocol
IP
b) TCMTF
*draft-saldana-tsvwg-tcmtf-05
Can We Multiplex ACKs without Harming the Performance of TCP? - CCNC 2014
8. Introduction
TCP video traffic: 69% of all consumer Internet traffic
in 2017.
When downloading a video, a computer may
generate some hundreds of ACKs per second, during
some tens of seconds.
In some scenarios (e.g. the aggregation network of an
operator) high numbers of long-term flows of ACKs
share a common path.
Header compression ratio of ACKs: from 40 to 7 or 8
bytes (savings of 80%).
Can We Multiplex ACKs without Harming the Performance of TCP? - CCNC 2014
9. Introduction
Is it a good idea to compress and multiplex these
flows?
Would the multiplexing delay degrade the
performance of TCP?
Sawtooth-shaped delay
PE
Added
delay
PE
time
Can We Multiplex ACKs without Harming the Performance of TCP? - CCNC 2014
10. Index
1. Introduction
2. Tests and results
3. Conclusions
Can We Multiplex ACKs without Harming the Performance of TCP? - CCNC 2014
11. Tests and Results
Dumbbell scenario in ns2
A sawtooth-shaped delay is added to the ACKs B-B’
A
O
ACK
FTP
N
B
P
ACK mux (PE)
A’
M
FTP
B’
What is the effect? We use TCP Tahoe (the most
basic one) in order to more clearly see the effect
First tests: separate A-A’ and B-B’
Can We Multiplex ACKs without Harming the Performance of TCP? - CCNC 2014
12. Tests and Results
Bandwidth [Mbps]
Throughput (RTT = 80 ms)
10
8
6
4
2
0
30
35
40
45
50
55
60
55
60
simulated time [s]
Bandwidth [Mbps]
Throughput (RTT = 80 ms, mux period 50 ms)
10
8
6
4
2
0
30
35
40
45
50
simulated time [s]
Can We Multiplex ACKs without Harming the Performance of TCP? - CCNC 2014
13. Tests and Results
Bandwidth [Mbps]
Throughput (RTT = 80 ms)
10
8
6
avg 9.24 Mbps
window reset
every ~7 sec
4
2
0
30
35
40
45
50
55
60
simulated time [s]
Bandwidth [Mbps]
ACKs arrive
Throughput (RTT = 80 ms, mux period 50 ms) in bursts
10
8
6
window reset
every ~9,5 sec
4
2
avg 8.04 Mbps
(12% reduction)
0
30
35
40
45
50
simulated time [s]
Can We Multiplex ACKs without Harming the Performance of TCP? - CCNC 2014
55
60
14. Tests and Results
Window size
250
no PE
PE=50ms
Window size
200
150
100
50
0
0
10
20
30
40
50
60
simulation time [s]
Can We Multiplex ACKs without Harming the Performance of TCP? - CCNC 2014
70
80
15. Tests and Results
Window evolution. One period
140
no PE
PE=50ms
120
Window size
100
Slow start
ends later
80
60
Window size
increases
more slowly
40
20
0
Can We Multiplex ACKs without Harming the Performance of TCP? - CCNC 2014
The period
between
window
resets
increases
16. Tests and Results
Second tests: A-A’ and B-B’ share the bottleneck
Are multiplexed flows in clear disadvantage?
We will use four different TCP variants:
Tahoe
FTP
A
O ACK
A’
Reno
N
M
New Reno B
FTP
P ACK mux (PE)
B’
SACK
Results: Throughput difference between
multiplexed and non-multiplexed flows
Can We Multiplex ACKs without Harming the Performance of TCP? - CCNC 2014
17. Tests and Results
Results: Throughput difference between
multiplexed and non-multiplexed flows
Multiplexing Period PE [ms]
TCP
5
10
15
20
25
Tahoe
4.91 %
10.05 %
31.67 %
7.88 %
49.74 %
Reno
5.95 %
17.78 %
48.62 %
24.29 %
61.92 %
New
Reno
4.82 %
12.95 %
30.52 %
16.70 %
52.87 %
SACK
2.27 %
12.70 %
20.62 %
14.75 %
50.90 %
Can We Multiplex ACKs without Harming the Performance of TCP? - CCNC 2014
18. Tests and Results
Results: PE=5 ms
With Throughput difference between
the difference
multiplexed and non-multiplexed flows
is small
Multiplexing Period PE [ms]
TCP
5
10
15
20
25
Tahoe
4.91 %
10.05 %
31.67 %
7.88 %
49.74 %
Reno
5.95 %
17.78 %
48.62 %
24.29 %
61.92 %
New
Reno
4.82 %
12.95 %
30.52 %
16.70 %
52.87 %
SACK
2.27 %
12.70 %
20.62 %
14.75 %
50.90 %
Can We Multiplex ACKs without Harming the Performance of TCP? - CCNC 2014
19. Tests and Results
Results: Throughputms
With PE=10 difference between
the difference
multiplexed and non-multiplexed flows
becomes higher
Multiplexing Period PE [ms]
TCP
5
10
15
20
25
Tahoe
4.91 %
10.05 %
31.67 %
7.88 %
49.74 %
Reno
5.95 %
17.78 %
48.62 %
24.29 %
61.92 %
New
Reno
4.82 %
12.95 %
30.52 %
16.70 %
52.87 %
SACK
2.27 %
12.70 %
20.62 %
14.75 %
50.90 %
Can We Multiplex ACKs without Harming the Performance of TCP? - CCNC 2014
20. Tests and Results
Results: Throughput difference between
Below 10 ms the
difference may
multiplexed and non-multiplexed flows
become
unacceptable
Multiplexing Period PE [ms]
TCP
5
10
15
20
25
Tahoe
4.91 %
10.05 %
31.67 %
7.88 %
49.74 %
Reno
5.95 %
17.78 %
48.62 %
24.29 %
61.92 %
New
Reno
4.82 %
12.95 %
30.52 %
16.70 %
52.87 %
SACK
2.27 %
12.70 %
20.62 %
14.75 %
50.90 %
Can We Multiplex ACKs without Harming the Performance of TCP? - CCNC 2014
21. Tests and Results
Throughput (SACK)
no PE
PE=5 ms
10
Throughput [Mbps]
8
6
4
2
0
900
910
920
930
940
950
960
970
980
Simulation time
Can We Multiplex ACKs without Harming the Performance of TCP? - CCNC 2014
990
1000
22. Tests and Results
Throughput (Reno)
no PE
PE=25 ms
10
Throughput [Mbps]
8
6
4
2
0
900
910
920
930
940
950
960
970
980
Simulation time
Can We Multiplex ACKs without Harming the Performance of TCP? - CCNC 2014
990
1000
23. Index
1. Introduction
2. Tests and results
3. Conclusions
Can We Multiplex ACKs without Harming the Performance of TCP? - CCNC 2014
24. Conclusions
Suitability of traffic optimization, based on header
compression and multiplexing, to the flows of ACKs
The expected bandwidth savings are huge because of
the absence of payload
Counterpart: throughput reduction when an
optimized flow shares a bottleneck with a nonoptimized one
The impairments can be maintained in tolerable
limits, by setting an upper bound on the period
Further study this trade-off between bandwidth
saving and throughput reduction
Can We Multiplex ACKs without Harming the Performance of TCP? - CCNC 2014
25. Thank you very much!
Jose Saldana, Julián Fernández-Navajas, José Ruiz-Mas