This technical whitepaper compares Aspera FASP, a high-speed transport protocol, to alternative TCP-based and UDP-based file transfer technologies. It finds that while TCP and high-speed TCP variants can improve throughput over standard TCP in low-loss networks, their performance degrades significantly in wide-area networks with higher latency and packet loss. UDP-based solutions also struggle to achieve high throughput and efficiency across different network conditions due to poor congestion control. In contrast, Aspera FASP is able to achieve maximum throughput that is independent of network characteristics like latency and packet loss, making it optimal for reliable, high-speed transfer of large files over IP networks.
Sky X products provide performance enhancement for data transmissions over satellite networks by replacing TCP with a custom protocol called Sky X that is optimized for satellite conditions like long latency and high bit error rates. The Sky X Gateway intercepts TCP connections and converts the data to the Sky X protocol for transmission over the satellite. This solution increases web and file transfer speeds by 3 to 100 times compared to TCP over satellite. The Sky X products transparently replace TCP and do not require any client or server modifications.
Iaetsd an effective approach to eliminate tcp incastIaetsd Iaetsd
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
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.
PERFORMANCE EVALUATION OF SELECTED E2E TCP CONGESTION CONTROL MECHANISM OVER ...ijwmn
TCP is one of the main protocols that govern the Internet traffic nowadays. However, it suffers significant
performance degradation over wireless links. Since wireless networks are leading the communication
technologies recently, it is imperative to introduce effective solutions for the TCP congestion control
mechanisms over such networks. In this research four End-to-End TCP implementations are discussed,
they are TCP Westwood, Hybla, Highspeed, and NewReno. The performance of these variants is compared
using LTE emulated environment in terms of throughput, delay, and fairness. Ns-3 simulator is used to
simulate the LTE networks environment. The simulation results showed that TCP Highspeed achieves the
best throughput results. Although TCP Westwood recorded the lowest latency values comparing to others,
it behaved unfairly among different traffic flows. Moreover, TCP Hybla demonstrated the best fairness
behaviour among other TCP variants
International Journal of Engineering Research and Applications (IJERA) is a team of researchers not publication services or private publications running the journals for monetary benefits, we are association of scientists and academia who focus only on supporting authors who want to publish their work. The articles published in our journal can be accessed online, all the articles will be archived for real time access.
Our journal system primarily aims to bring out the research talent and the works done by sciaentists, academia, engineers, practitioners, scholars, post graduate students of engineering and science. This journal aims to cover the scientific research in a broader sense and not publishing a niche area of research facilitating researchers from various verticals to publish their papers. It is also aimed to provide a platform for the researchers to publish in a shorter of time, enabling them to continue further All articles published are freely available to scientific researchers in the Government agencies,educators and the general public. We are taking serious efforts to promote our journal across the globe in various ways, we are sure that our journal will act as a scientific platform for all researchers to publish their works online.
Dear Students
Ingenious techno Solution offers an expertise guidance on you Final Year IEEE & Non- IEEE Projects on the following domain
JAVA
.NET
EMBEDDED SYSTEMS
ROBOTICS
MECHANICAL
MATLAB etc
For further details contact us:
enquiry@ingenioustech.in
044-42046028 or 8428302179.
Ingenious Techno Solution
#241/85, 4th floor
Rangarajapuram main road,
Kodambakkam (Power House)
http://www.ingenioustech.in/
TCP Santa Cruz is a new implementation of TCP congestion control and error recovery designed to work better than TCP Reno or Tahoe over networks with heterogeneous transmission media. It uses estimates of the relative delay between packets on the forward path, rather than round-trip time estimates, to detect congestion early. It can identify the direction of congestion to isolate the forward throughput from reverse path events. Simulation experiments show TCP Santa Cruz achieves significantly higher throughput, smaller delays, and delay variances than TCP Reno and Vegas.
Performance Evaluation of UDP, DCCP, SCTP and TFRC for Different Traffic Flow...IJECEIAES
The demand for internet applications has increased rapidly. Providing quality of service (QoS) requirements for varied internet application is a challenging task. One important factor that is significantly affected on the QoS service is the transport layer. The transport layer provides end-to-end data transmission across a network. Currently, the most common transport protocols used by internet application are TCP (Transmission Control Protocol) and UDP (User Datagram Protocol). Also, there are recent transport protocols such as DCCP (data congestion control protocol), SCTP (stream congestion transmission protocol), and TFRC (TCP-friendly rate control), which are in the standardization process of Internet Engineering Task Force (IETF). In this paper, we evaluate the performance of UDP, DCCP, SCTP and TFRC protocols for different traffic flows: data transmission, video traffic, and VOIP in wired networks. The performance criteria used for this evaluation include throughput, end to end delay, and packet loss rate. Well-known network simulator NS-2 used to implement the UDP, DCCP, SCTP, and TFRC protocols performance comparison. Based on the simulation results, the performance throughput of SCTP and TFRC is better than UDP. Moreover, DCCP performance is superior SCTP and TFRC in term of end-to-end delay.
Sky X products provide performance enhancement for data transmissions over satellite networks by replacing TCP with a custom protocol called Sky X that is optimized for satellite conditions like long latency and high bit error rates. The Sky X Gateway intercepts TCP connections and converts the data to the Sky X protocol for transmission over the satellite. This solution increases web and file transfer speeds by 3 to 100 times compared to TCP over satellite. The Sky X products transparently replace TCP and do not require any client or server modifications.
Iaetsd an effective approach to eliminate tcp incastIaetsd Iaetsd
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
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.
PERFORMANCE EVALUATION OF SELECTED E2E TCP CONGESTION CONTROL MECHANISM OVER ...ijwmn
TCP is one of the main protocols that govern the Internet traffic nowadays. However, it suffers significant
performance degradation over wireless links. Since wireless networks are leading the communication
technologies recently, it is imperative to introduce effective solutions for the TCP congestion control
mechanisms over such networks. In this research four End-to-End TCP implementations are discussed,
they are TCP Westwood, Hybla, Highspeed, and NewReno. The performance of these variants is compared
using LTE emulated environment in terms of throughput, delay, and fairness. Ns-3 simulator is used to
simulate the LTE networks environment. The simulation results showed that TCP Highspeed achieves the
best throughput results. Although TCP Westwood recorded the lowest latency values comparing to others,
it behaved unfairly among different traffic flows. Moreover, TCP Hybla demonstrated the best fairness
behaviour among other TCP variants
International Journal of Engineering Research and Applications (IJERA) is a team of researchers not publication services or private publications running the journals for monetary benefits, we are association of scientists and academia who focus only on supporting authors who want to publish their work. The articles published in our journal can be accessed online, all the articles will be archived for real time access.
Our journal system primarily aims to bring out the research talent and the works done by sciaentists, academia, engineers, practitioners, scholars, post graduate students of engineering and science. This journal aims to cover the scientific research in a broader sense and not publishing a niche area of research facilitating researchers from various verticals to publish their papers. It is also aimed to provide a platform for the researchers to publish in a shorter of time, enabling them to continue further All articles published are freely available to scientific researchers in the Government agencies,educators and the general public. We are taking serious efforts to promote our journal across the globe in various ways, we are sure that our journal will act as a scientific platform for all researchers to publish their works online.
Dear Students
Ingenious techno Solution offers an expertise guidance on you Final Year IEEE & Non- IEEE Projects on the following domain
JAVA
.NET
EMBEDDED SYSTEMS
ROBOTICS
MECHANICAL
MATLAB etc
For further details contact us:
enquiry@ingenioustech.in
044-42046028 or 8428302179.
Ingenious Techno Solution
#241/85, 4th floor
Rangarajapuram main road,
Kodambakkam (Power House)
http://www.ingenioustech.in/
TCP Santa Cruz is a new implementation of TCP congestion control and error recovery designed to work better than TCP Reno or Tahoe over networks with heterogeneous transmission media. It uses estimates of the relative delay between packets on the forward path, rather than round-trip time estimates, to detect congestion early. It can identify the direction of congestion to isolate the forward throughput from reverse path events. Simulation experiments show TCP Santa Cruz achieves significantly higher throughput, smaller delays, and delay variances than TCP Reno and Vegas.
Performance Evaluation of UDP, DCCP, SCTP and TFRC for Different Traffic Flow...IJECEIAES
The demand for internet applications has increased rapidly. Providing quality of service (QoS) requirements for varied internet application is a challenging task. One important factor that is significantly affected on the QoS service is the transport layer. The transport layer provides end-to-end data transmission across a network. Currently, the most common transport protocols used by internet application are TCP (Transmission Control Protocol) and UDP (User Datagram Protocol). Also, there are recent transport protocols such as DCCP (data congestion control protocol), SCTP (stream congestion transmission protocol), and TFRC (TCP-friendly rate control), which are in the standardization process of Internet Engineering Task Force (IETF). In this paper, we evaluate the performance of UDP, DCCP, SCTP and TFRC protocols for different traffic flows: data transmission, video traffic, and VOIP in wired networks. The performance criteria used for this evaluation include throughput, end to end delay, and packet loss rate. Well-known network simulator NS-2 used to implement the UDP, DCCP, SCTP, and TFRC protocols performance comparison. Based on the simulation results, the performance throughput of SCTP and TFRC is better than UDP. Moreover, DCCP performance is superior SCTP and TFRC in term of end-to-end delay.
Module 3: Transport layer
Transport layer services, Multiplexing and demultiplexing, User datagram protocol, Transmission control protocol: connection, features, segment, Round-Trip Time estimation and timeout, Flow control, Congestion control, SCTP
A New Data Link Layer Protocolfor Satellite IP NetworksNiraj Solanki
NSLP is a new satellite link protocol proposed for satellite IP networks. It simplifies the data link layer function and uses a variable length frame format coupled to IP packet data. This leads to higher transmission efficiency, lower IP packet loss rates, and better compatibility with TCP/IP compared to other protocols like CCSDS and HDLC. Simulation results show NSLP can improve performance for satellite IP networks by reducing overhead and improving utilization of limited satellite resources.
This document discusses quality of service (QoS) techniques for managing bandwidth and latency requirements of different network applications like VoIP. It covers class of service and type of service fields which allow grouping of packet flows. It also discusses queuing techniques like weighted fair queuing, priority queuing, and custom queuing which allow controlling bandwidth allocation to different traffic types. Packet classification methods like IP precedence and policy routing are also covered which allow setting priority levels for traffic.
TCP INCAST AVOIDANCE BASED ON CONNECTION SERIALIZATION IN DATA CENTER NETWORKSIJCNCJournal
In distributed file systems, a well-known congestion collapse called TCP incast (Incast briefly) occurs
because many servers almost simultaneously send data to the same client and then many packets overflow
the port buffer of the link connecting to the client. Incast leads to throughput degradation in the network. In
this paper, we propose three methods to avoid Incast based on the fact that the bandwidth-delay product is
small in current data center networks. The first method is a method which completely serializes connection
establishments. By the serialization, the number of packets in the port buffer becomes very small, which
leads to Incast avoidance. The second and third methods are methods which overlap the slow start period
of the next connection with the current established connection to improve throughput in the first method.
Numerical results from extensive simulation runs show the effectiveness of our three proposed methods.
Study on Performance of Simulation Analysis on Multimedia NetworkIRJET Journal
This document summarizes a study that simulated voice communication over wired networks using the NS-2 network simulator. The study modeled VoIP traffic between nodes using the SCTP protocol and added background traffic to evaluate its effects. Key findings from the simulation included:
1) Average latency was 0.98 seconds and 98 packets were dropped, indicating degraded performance when background traffic was added.
2) Average jitter (packet delay variation) was calculated to be 0.006 seconds, showing instability in the network with changing traffic patterns.
3) A graph of latency over time demonstrated increased delays and bottlenecks as background traffic overloaded network links.
This document summarizes key concepts from Chapter 3 of the textbook on transport layer protocols:
1. The transport layer provides logical communication between processes running on different hosts, abstracting the underlying network infrastructure. It multiplexes data from multiple sockets and demultiplexes received data to the appropriate socket.
2. UDP and TCP are the main transport protocols in the Internet. UDP is connectionless while TCP provides reliable, connection-oriented data transfer using sequence numbers, acknowledgments, and congestion control.
3. TCP uses congestion control including a congestion window, additive increase/multiplicative decrease, and slow start to dynamically control the sender's transmission rate based on detected packet loss as a signal of
A COMPARISON OF CONGESTION CONTROL VARIANTS OF TCP IN REACTIVE ROUTING PROTOC...ijcsit
A widely used TCP protocol is originally developed for wired networks. It has many variants to detect and
control congestion in the network. However, Congestion control in all TCP variants does not show similar
performance in MANET as in wired network because of the fault detection of congestion. In this paper, we
do a performance comparison between TCP variants NEW RENO, SACK and Vegas in AODV and DSR
reactive (On-Demand) routing protocols. Network traffic between nodes is provided by using File Transfer
Protocol (FTP) application. Multiple scenarios are created and the average values of each performance
parameter are used to evaluate the performance. The results show that TCP variants perform better in
terms of throughput and Packet drop with DSR routing protocol compared with AODV routing protocol.
TCP variants show a lower Jitter in AODV compared with DSR.
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.
This document discusses integrated services architecture (ISA) and differentiated services (DS) for providing quality of service (QoS) in computer networks. It describes the components and functions of ISA, including reservation protocol, admission control, routing, queuing disciplines, and services. It also covers traffic classification, scheduling, and dropping policies implemented in routers. Random early detection (RED) is presented as a proactive packet discard mechanism for congestion management. Differentiated services is introduced as a simpler alternative to ISA that uses traffic classes in packet headers to provide different performance levels.
This document discusses TCP flow and congestion control in high speed networks. It covers topics such as TCP flow control using a credit allocation scheme, TCP header fields for flow control, credit allocation flexibility, effects of window size, complicating factors, retransmission strategy using timers, adaptive retransmission timer algorithms, implementation policy options, congestion control difficulties, slow start, dynamic window sizing, fast retransmit, fast recovery, limited transmit, performance of TCP over ATM networks using UBR service, effects of switch buffer size, observations, partial packet discard techniques, and TCP over ABR service.
This document provides an overview of the transport layer chapter from the textbook "Computer Networking: A Top Down Approach". It discusses the goals of understanding transport layer services like multiplexing and demultiplexing. It also covers the two main Internet transport protocols - UDP which provides connectionless unreliable data transfer, and TCP which provides connection-oriented reliable data transfer. The document outlines the rest of the chapter which will discuss TCP and UDP in more detail as well as principles of reliable data transfer and congestion control.
Transport Layer Services : Multiplexing And DemultiplexingKeyur Vadodariya
This document discusses the transport layer of computer networks. It begins with introducing the group members and topic, which is the transport layer introduction, services, multiplexing and demultiplexing. Then it provides definitions of the transport layer, its functions and services. It describes how the transport layer provides process to process delivery, end-to-end connections, congestion control, data integrity, flow control, multiplexing and demultiplexing. It explains the differences between connectionless and connection-oriented multiplexing and demultiplexing. In the end, it lists some references.
1. The document discusses the WAP (Wireless Application Protocol) architecture and its components for enabling wireless internet access on mobile devices. It includes protocols like WDP, WTLS, WSP, and WML.
2. The WAP architecture consists of a transport layer, security layer, transaction layer, session layer, and application layer. It maps to internet protocols like TCP/IP, TLS, and HTTP to provide similar functionality to mobile devices.
3. Special adaptations were required for the wireless environment, including new protocols like WML, a binary version of HTML, and WTA for wireless telephony integration. Gateways translate between internet protocols and WAP to enable access of web and other internet content on mobile
T/TCP is a protocol that aims to reduce the number of packets needed for transaction-style applications by allowing a client to open a connection, send data, and close the connection in a single packet. It utilizes a mechanism called TCP Accelerated Open (TAO) to bypass the standard 3-way TCP handshake. Testing showed T/TCP saved an average of 5 packets per transaction compared to TCP. However, the percentage savings decreased with larger data transfers as T/TCP is most beneficial for small transactions. While improving performance, T/TCP also introduced some security and operational issues that needed to be addressed for broader adoption.
In this video you will learn
- Explain the need for the Transport layer.
- Identify the role of the Transport layer as it provides the end-to-end transfer of data between applications.
- Describe the role of two TCP/IP Transport layer protocols: TCP and UDP.
- Explain the key functions of the Transport layer, including reliability, port addressing, and segmentation.
- Explain how TCP and UDP each handle key functions.
- Identify when it is appropriate to use TCP or UDP and provide examples of applications that use each protocol.
avoiding retransmissions using random coding scheme or fountain code schemeIJAEMSJORNAL
In a perfect world, the throughput of a Multipath TCP (MPTCP) association ought to be as high as that of different disjoint single-way TCP streams. In actuality, the throughput of MPTCP is far lower than anticipated. In this paper, we lead angeneral reproduction construct ponder in light of this peculiarity, and the outcomes show that a sub stream encountering high postponement and misfortune extremely influences the execution of other sub streams, in this manner turning into the half back of the MPTCP association and knowingly humiliating the total great put. To handle this issue, we propose Wellspring code-based Multipath TCP (FMTCP), which viably mitigates the negative effect of the heterogeneity of modified ways. FMTCP exploits the unintentional way of the wellspring code to adaptably transmit encoded images from the same or diverse information hinders over various sub streams. In addition, we plan an information portion calculation in view of the foreseen bundle arriving time and deciphering command to facilitate the transmissions of various sub streams. Quantitative reviews are given to demonstrate the advantage of FMTCP. We likewise assess the presentation of FMTCP through ns-2 recreations and exhibit that FMTCP beats IETF-MPTCP, a run of the mill MPTCP approach, when the ways have various misfortune and deferral as far as higher aggregate great put, bring down postponement, and jitter. Also, FMTCP acknowledges high security under sudden changes of way quality.
The document discusses congestion control in computer networks. It defines congestion as occurring when the load on a network is greater than the network's capacity. Congestion control aims to control congestion and keep the load below capacity. The document outlines two categories of congestion control: open-loop control, which aims to prevent congestion; and closed-loop control, which detects congestion and takes corrective action using feedback from the network. Specific open-loop techniques discussed include admission control, traffic shaping using leaky bucket and token bucket algorithms, and traffic scheduling.
Assessing Buffering with Scheduling Schemes in a QoS Internet RouterIOSR Journals
This document examines different scheduling algorithms that could be used with RIO-C penalty enforcement buffering in a multi-queue QoS router to improve network performance. It simulates priority, round robin, and weighted round robin scheduling with RIO-C buffering. The results show that priority scheduling achieved the lowest loss rates, with 29.46% scheduler drop rate and 14.95% RED loss rate. Round robin was second with 29.53% and 10.50% losses. Weighted round robin was third with 30.28% and 3.04% losses. The document concludes that a network seeking quality of service could adopt priority scheduling with RIO-C admission control.
Aspera for Microsoft SharePoint is for organizations who need to quickly, predictably, and securely store and access high volumes of large files in Share Point. Aspera has seamlessly integrated patented FASP transfer technology into SharePoint for high-speed document upload and download workflows. Using Aspera, customers can not only overcome file and total repository size limitations of SharePoint, they also can reliably transfer files into and out of SharePoint at much higher speeds with auditable and predictable results. That means users can now upload large video files, imagery files, laser scan data, and other large files leveraging SharePoint document library structures and metadata for organization and search. Watch the Aspera for SharePoint Webinar at https://youtu.be/J8BOFBK-glc
FASP is a file transfer technology that is hundreds of times faster than FTP and HTTP. It guarantees delivery times regardless of file size or network conditions. FASP achieves maximum speeds by fully utilizing available bandwidth along the entire transfer path. Unlike TCP, FASP is not slowed down by network latency and packet loss. It provides secure transfers with authentication, encryption and integrity verification without compromising speed.
This document discusses enabling the effective sharing of medical images over wide area and wireless networks. It introduces Aspera, a company that creates transport technologies to move digital assets at maximum speed regardless of file size or network conditions. Specific healthcare applications are discussed, including transferring images between PACS systems, enabling health information exchange, and providing on-demand access to images. Potential integrations with organizations like Johns Hopkins and McKesson are also mentioned.
Module 3: Transport layer
Transport layer services, Multiplexing and demultiplexing, User datagram protocol, Transmission control protocol: connection, features, segment, Round-Trip Time estimation and timeout, Flow control, Congestion control, SCTP
A New Data Link Layer Protocolfor Satellite IP NetworksNiraj Solanki
NSLP is a new satellite link protocol proposed for satellite IP networks. It simplifies the data link layer function and uses a variable length frame format coupled to IP packet data. This leads to higher transmission efficiency, lower IP packet loss rates, and better compatibility with TCP/IP compared to other protocols like CCSDS and HDLC. Simulation results show NSLP can improve performance for satellite IP networks by reducing overhead and improving utilization of limited satellite resources.
This document discusses quality of service (QoS) techniques for managing bandwidth and latency requirements of different network applications like VoIP. It covers class of service and type of service fields which allow grouping of packet flows. It also discusses queuing techniques like weighted fair queuing, priority queuing, and custom queuing which allow controlling bandwidth allocation to different traffic types. Packet classification methods like IP precedence and policy routing are also covered which allow setting priority levels for traffic.
TCP INCAST AVOIDANCE BASED ON CONNECTION SERIALIZATION IN DATA CENTER NETWORKSIJCNCJournal
In distributed file systems, a well-known congestion collapse called TCP incast (Incast briefly) occurs
because many servers almost simultaneously send data to the same client and then many packets overflow
the port buffer of the link connecting to the client. Incast leads to throughput degradation in the network. In
this paper, we propose three methods to avoid Incast based on the fact that the bandwidth-delay product is
small in current data center networks. The first method is a method which completely serializes connection
establishments. By the serialization, the number of packets in the port buffer becomes very small, which
leads to Incast avoidance. The second and third methods are methods which overlap the slow start period
of the next connection with the current established connection to improve throughput in the first method.
Numerical results from extensive simulation runs show the effectiveness of our three proposed methods.
Study on Performance of Simulation Analysis on Multimedia NetworkIRJET Journal
This document summarizes a study that simulated voice communication over wired networks using the NS-2 network simulator. The study modeled VoIP traffic between nodes using the SCTP protocol and added background traffic to evaluate its effects. Key findings from the simulation included:
1) Average latency was 0.98 seconds and 98 packets were dropped, indicating degraded performance when background traffic was added.
2) Average jitter (packet delay variation) was calculated to be 0.006 seconds, showing instability in the network with changing traffic patterns.
3) A graph of latency over time demonstrated increased delays and bottlenecks as background traffic overloaded network links.
This document summarizes key concepts from Chapter 3 of the textbook on transport layer protocols:
1. The transport layer provides logical communication between processes running on different hosts, abstracting the underlying network infrastructure. It multiplexes data from multiple sockets and demultiplexes received data to the appropriate socket.
2. UDP and TCP are the main transport protocols in the Internet. UDP is connectionless while TCP provides reliable, connection-oriented data transfer using sequence numbers, acknowledgments, and congestion control.
3. TCP uses congestion control including a congestion window, additive increase/multiplicative decrease, and slow start to dynamically control the sender's transmission rate based on detected packet loss as a signal of
A COMPARISON OF CONGESTION CONTROL VARIANTS OF TCP IN REACTIVE ROUTING PROTOC...ijcsit
A widely used TCP protocol is originally developed for wired networks. It has many variants to detect and
control congestion in the network. However, Congestion control in all TCP variants does not show similar
performance in MANET as in wired network because of the fault detection of congestion. In this paper, we
do a performance comparison between TCP variants NEW RENO, SACK and Vegas in AODV and DSR
reactive (On-Demand) routing protocols. Network traffic between nodes is provided by using File Transfer
Protocol (FTP) application. Multiple scenarios are created and the average values of each performance
parameter are used to evaluate the performance. The results show that TCP variants perform better in
terms of throughput and Packet drop with DSR routing protocol compared with AODV routing protocol.
TCP variants show a lower Jitter in AODV compared with DSR.
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.
This document discusses integrated services architecture (ISA) and differentiated services (DS) for providing quality of service (QoS) in computer networks. It describes the components and functions of ISA, including reservation protocol, admission control, routing, queuing disciplines, and services. It also covers traffic classification, scheduling, and dropping policies implemented in routers. Random early detection (RED) is presented as a proactive packet discard mechanism for congestion management. Differentiated services is introduced as a simpler alternative to ISA that uses traffic classes in packet headers to provide different performance levels.
This document discusses TCP flow and congestion control in high speed networks. It covers topics such as TCP flow control using a credit allocation scheme, TCP header fields for flow control, credit allocation flexibility, effects of window size, complicating factors, retransmission strategy using timers, adaptive retransmission timer algorithms, implementation policy options, congestion control difficulties, slow start, dynamic window sizing, fast retransmit, fast recovery, limited transmit, performance of TCP over ATM networks using UBR service, effects of switch buffer size, observations, partial packet discard techniques, and TCP over ABR service.
This document provides an overview of the transport layer chapter from the textbook "Computer Networking: A Top Down Approach". It discusses the goals of understanding transport layer services like multiplexing and demultiplexing. It also covers the two main Internet transport protocols - UDP which provides connectionless unreliable data transfer, and TCP which provides connection-oriented reliable data transfer. The document outlines the rest of the chapter which will discuss TCP and UDP in more detail as well as principles of reliable data transfer and congestion control.
Transport Layer Services : Multiplexing And DemultiplexingKeyur Vadodariya
This document discusses the transport layer of computer networks. It begins with introducing the group members and topic, which is the transport layer introduction, services, multiplexing and demultiplexing. Then it provides definitions of the transport layer, its functions and services. It describes how the transport layer provides process to process delivery, end-to-end connections, congestion control, data integrity, flow control, multiplexing and demultiplexing. It explains the differences between connectionless and connection-oriented multiplexing and demultiplexing. In the end, it lists some references.
1. The document discusses the WAP (Wireless Application Protocol) architecture and its components for enabling wireless internet access on mobile devices. It includes protocols like WDP, WTLS, WSP, and WML.
2. The WAP architecture consists of a transport layer, security layer, transaction layer, session layer, and application layer. It maps to internet protocols like TCP/IP, TLS, and HTTP to provide similar functionality to mobile devices.
3. Special adaptations were required for the wireless environment, including new protocols like WML, a binary version of HTML, and WTA for wireless telephony integration. Gateways translate between internet protocols and WAP to enable access of web and other internet content on mobile
T/TCP is a protocol that aims to reduce the number of packets needed for transaction-style applications by allowing a client to open a connection, send data, and close the connection in a single packet. It utilizes a mechanism called TCP Accelerated Open (TAO) to bypass the standard 3-way TCP handshake. Testing showed T/TCP saved an average of 5 packets per transaction compared to TCP. However, the percentage savings decreased with larger data transfers as T/TCP is most beneficial for small transactions. While improving performance, T/TCP also introduced some security and operational issues that needed to be addressed for broader adoption.
In this video you will learn
- Explain the need for the Transport layer.
- Identify the role of the Transport layer as it provides the end-to-end transfer of data between applications.
- Describe the role of two TCP/IP Transport layer protocols: TCP and UDP.
- Explain the key functions of the Transport layer, including reliability, port addressing, and segmentation.
- Explain how TCP and UDP each handle key functions.
- Identify when it is appropriate to use TCP or UDP and provide examples of applications that use each protocol.
avoiding retransmissions using random coding scheme or fountain code schemeIJAEMSJORNAL
In a perfect world, the throughput of a Multipath TCP (MPTCP) association ought to be as high as that of different disjoint single-way TCP streams. In actuality, the throughput of MPTCP is far lower than anticipated. In this paper, we lead angeneral reproduction construct ponder in light of this peculiarity, and the outcomes show that a sub stream encountering high postponement and misfortune extremely influences the execution of other sub streams, in this manner turning into the half back of the MPTCP association and knowingly humiliating the total great put. To handle this issue, we propose Wellspring code-based Multipath TCP (FMTCP), which viably mitigates the negative effect of the heterogeneity of modified ways. FMTCP exploits the unintentional way of the wellspring code to adaptably transmit encoded images from the same or diverse information hinders over various sub streams. In addition, we plan an information portion calculation in view of the foreseen bundle arriving time and deciphering command to facilitate the transmissions of various sub streams. Quantitative reviews are given to demonstrate the advantage of FMTCP. We likewise assess the presentation of FMTCP through ns-2 recreations and exhibit that FMTCP beats IETF-MPTCP, a run of the mill MPTCP approach, when the ways have various misfortune and deferral as far as higher aggregate great put, bring down postponement, and jitter. Also, FMTCP acknowledges high security under sudden changes of way quality.
The document discusses congestion control in computer networks. It defines congestion as occurring when the load on a network is greater than the network's capacity. Congestion control aims to control congestion and keep the load below capacity. The document outlines two categories of congestion control: open-loop control, which aims to prevent congestion; and closed-loop control, which detects congestion and takes corrective action using feedback from the network. Specific open-loop techniques discussed include admission control, traffic shaping using leaky bucket and token bucket algorithms, and traffic scheduling.
Assessing Buffering with Scheduling Schemes in a QoS Internet RouterIOSR Journals
This document examines different scheduling algorithms that could be used with RIO-C penalty enforcement buffering in a multi-queue QoS router to improve network performance. It simulates priority, round robin, and weighted round robin scheduling with RIO-C buffering. The results show that priority scheduling achieved the lowest loss rates, with 29.46% scheduler drop rate and 14.95% RED loss rate. Round robin was second with 29.53% and 10.50% losses. Weighted round robin was third with 30.28% and 3.04% losses. The document concludes that a network seeking quality of service could adopt priority scheduling with RIO-C admission control.
Aspera for Microsoft SharePoint is for organizations who need to quickly, predictably, and securely store and access high volumes of large files in Share Point. Aspera has seamlessly integrated patented FASP transfer technology into SharePoint for high-speed document upload and download workflows. Using Aspera, customers can not only overcome file and total repository size limitations of SharePoint, they also can reliably transfer files into and out of SharePoint at much higher speeds with auditable and predictable results. That means users can now upload large video files, imagery files, laser scan data, and other large files leveraging SharePoint document library structures and metadata for organization and search. Watch the Aspera for SharePoint Webinar at https://youtu.be/J8BOFBK-glc
FASP is a file transfer technology that is hundreds of times faster than FTP and HTTP. It guarantees delivery times regardless of file size or network conditions. FASP achieves maximum speeds by fully utilizing available bandwidth along the entire transfer path. Unlike TCP, FASP is not slowed down by network latency and packet loss. It provides secure transfers with authentication, encryption and integrity verification without compromising speed.
This document discusses enabling the effective sharing of medical images over wide area and wireless networks. It introduces Aspera, a company that creates transport technologies to move digital assets at maximum speed regardless of file size or network conditions. Specific healthcare applications are discussed, including transferring images between PACS systems, enabling health information exchange, and providing on-demand access to images. Potential integrations with organizations like Johns Hopkins and McKesson are also mentioned.
This short document promotes the creation of Haiku Deck presentations on SlideShare and encourages the reader to get started making their own. It contains 5 stock photos credited to different photographers and writers. In under 3 sentences, it highlights Haiku Deck and SlideShare for making presentations and calls the reader to action to create their own.
Terri Moser is seeking employment as a Marriage and Family Therapist in Arizona. She has a Master's degree in Marriage and Family Therapy from Argosy University and is licensed in Utah. Her resume details over 15 years of experience providing therapy and counseling services to individuals, families, couples and groups through various treatment centers, focusing on addiction and mental health. She is a certified instructor in parenting and relationship skills.
This document summarizes a case study on the emerging role of school leaders in Pakistan as security administrators in the post-9/11 era. It finds that after 9/11, security became a top priority for school administration in Pakistan. It identifies key actors and factors important for school security, including trained school administrators. However, it also finds gaps in implementing security plans. It recommends further research and strengthening the role of school administrators in security through collaboration with policymakers.
The Migration Engine from Butterfly Software is a software solution that facilitates the automated consolidation of historic compliance data from legacy backup environments onto a single backup platform. It utilizes intelligent automation to safely migrate all data files and attributes in a strictly defined process with complete risk mitigation. The migration process involves discovery of data to migrate, scheduling backups in batches according to priorities, configuring the target environment, securely transferring data batches to the new backup platform, and validating the data and decommissioning legacy systems once complete. The Migration Engine provides total control and scalability during the migration process to optimize storage capacity and quickly transfer data with zero risk.
Teppala vijaya kumar is applying for the position of REVIT MEP Design Engineer. He has over 5 years of experience in REVIT MEP design for various commercial, residential, hospital, and industrial projects. He is proficient in REVIT MEP, AutoCAD, Navisworks, and has experience coordinating MEP designs and managing projects. He holds a B.Tech in Mechanical Engineering and is looking for a challenging position where he can further develop his skills.
This document is an excerpt from the book "Cloud Services For Dummies, IBM Limited Edition" which provides an introduction to cloud computing concepts and services. It discusses foundational elements of cloud services including delivery models, capabilities, and the cloud continuum. It also explores infrastructure as a service (IaaS) and platform as a service (PaaS) models in more detail including characteristics, uses, and considerations for evaluating these services. Finally, it covers additional topics related to cloud adoption like workload management, security, governance, and developing a strategy for transitioning to cloud.
This document contains information about various design projects including magazine layouts, album covers, advertisements, and reports. It discusses enhancing photos with effects, designing for specific genres and audiences, and creating visual identities through integrated design. It also includes details of an advertising campaign for Pizza Capers including TV commercial scripts, mail distribution strategies, and advertising rates.
The secret to true patient centricity parke ipJeff Parke
1) For the past couple of years, patient centricity has become a major focus in the pharmaceutical industry but companies need to ensure it is more than just marketing.
2) The document discusses core principles of patient centricity such as regular contact with patients, understanding the patient perspective, and open innovation to meet patient needs.
3) It argues that pharmaceutical companies should focus on improving patient outcomes and regaining trust through a truly patient-centric approach in all aspects of their work.
The document describes the author's work experience from 2008-2015 managing demolition, remediation, and hazardous material removal projects for various industrial clients. It details 14 separate projects the author worked on, primarily in the Netherlands, involving the total or selective demolition of buildings and infrastructure. The author's roles included inspections, asbestos and chemical investigations, project management, and health and safety coordination.
Christopher Ferry is a highly numerate recent graduate seeking a position as a trainee actuary. He has a MSc in Operational Research from the University of Southampton with Distinction and a First Class Honours in Mathematics. He has work experience in compliance services and trust companies, developing skills in teamwork, client services, and data management. Ferry is proficient in various computer programs and statistical software packages relevant to actuarial work. He is available to start in September 2015.
This document discusses fast and secure protocols for data transmission. It begins by defining what a protocol is and provides examples like HTTP and FTP. It then explains TCP and how it enables connection and guaranteed delivery of packets in order. FASP is introduced as an innovative file transfer technology that can achieve speeds hundreds of times faster than TCP/HTTP with guaranteed delivery times regardless of file size or network conditions. Benefits of FASP include maximum speed, security, and less packet loss compared to TCP. FASP is concluded to be a next generation technology that can provide optimal data transfer over any network.
Microsoft RemoteFX promises to enhance the user QoE for rich media applications running on remote desktops and IPQ can be a key technology to help deliver on that promise.
This document summarizes a review paper on congestion control approaches for real-time streaming applications on the Internet. It discusses how TCP is not well-suited for real-time streaming due to its reliance on packet loss and variable bitrates. The paper reviews different end-to-end and active queue management approaches for congestion control that aim to reduce latency and jitter. It covers issues with single and shared bottlenecks on the Internet that can lead to congestion and the need for new transport protocols and congestion control for real-time media streaming.
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MainlineNet Holdings owns Extreme TCP, a new technology that improves upon the standard TCP congestion avoidance algorithm used on the internet. Extreme TCP uses complex router and network models to transmit data at higher speeds while avoiding congestion events that slow transmission. Testing showed transmission speed improvements of 400-1000% for some connections and up to 1400% for longer connections. While Extreme TCP can be applied as a software patch, its benefits are only seen on the device that has it installed, not requiring widespread adoption. It has the potential to significantly increase transmission speeds for most internet communications that use TCP.
Communication Performance Over A Gigabit Ethernet NetworkIJERA Editor
A present computing imposes heavy demands on the optical communication network. Gigabit Ethernet technology can provide the required bandwidth to meet these demands. However, it has also involve the communication Impediment to progress from network media to TCP(Transfer control protocol) processing. In this paper, present an overview of Gigabit per second Ethernet technology and study the end-to-end Gigabit Ethernet communication bandwidth and retrieval time. Performance graphs are collected using NetPipe in this clearly show the performance characteristics of TCP/IP over Gigabit Ethernet. These indicate the impact of a number of factors such as processor speeds, network adaptors, versions of the Linux Kernel or opnet softwar and device drivers, and TCP/IP(Internet protocol) tuning on the performance of Gigabit Ethernet between two Pentium II/350 PCs. Among the important conclusions are the marked superiority of the 2.1.121 and later development kernels and 2.2.x production kernels of Linux or opnet softwar used and that the ability to increase the MTU(maximum transmission unit) Further than the Ethernet standard of 1500 could significantly enhance the throughput reachable.
In the last few years, video streaming facilities over TCP or UDP, such as YouTube, Facetime, Daily-motion, Mobile video calling have become more and more popular. The important
challenge in streaming broadcasting over the Internet is to spread the uppermost potential quality,
observe to the broadcasting play out time limitation, and efficiently and equally share the offered
bandwidth with TCP or UDP, and additional traffic types. This work familiarizes the Streaming
Media Data Congestion Control protocol (SMDCC), a new adaptive broadcasting streaming
congestion management protocol in which the connection’s data packets transmission frequency is
adjusted allowing to the dynamic bandwidth share of connection using SMDCC, the bandwidth share
of a connection is projected using algorithms similar to those introduced in TCP Westwood. SMDCC
avoids the Slow Jump phase in TCP. As a result, SMDCC does not show the pronounced rate
alternations distinguishing of modern TCP, so providing congestion control that is more appropriate
for streaming broadcasting applications. Besides, SMDCC is fair, sharing the bandwidth equitably
among a set of SMDCC connections. Main benefit is robustness when packet harms are due to
indiscriminate errors, which is typical of wireless links and is becoming an increasing concern due to
the emergence of wireless Internet access. In the presence of indiscriminate errors, SMDCC is also
approachable to TCP Tahoe and Reno (TTR). We provide simulation results using the ns3 simulator
for our protocol running together with TCP Tahoe and Reno.
The transport layer provides efficient, reliable, and cost-effective process-to-process delivery by making use of network layer services. The transport layer works through transport entities to achieve its goal of reliable delivery between application processes. It provides an interface for applications to access its services.
1 improvement of tcp congestion window over ltetanawan44
This document discusses improving the performance of TCP congestion control over LTE-Advanced networks. It proposes a new congestion avoidance mechanism that uses the available bandwidth of the connection to better detect the network path capacity and improve congestion avoidance. The mechanism is tested using the NS-2 network simulator to model LTE-Advanced traffic. The document provides background on LTE-Advanced network architecture and existing TCP congestion control mechanisms. It aims to develop an enhanced TCP variant that can efficiently transfer high data rates over the large bandwidth, low latency links of LTE-Advanced networks.
IP resides at the network layer and provides logical addressing that allows systems on different logical networks to communicate. It is a connectionless protocol that does not provide reliability, flow control, or sequencing. VoIP uses RTP, which sits atop UDP, to transport real-time voice data in an efficient manner without retransmissions. UDP is used instead of TCP for VoIP as reliability is less important than latency for real-time voice communications.
Performance Evaluation of TCP with Adaptive Pacing and LRED in Multihop Wirel...ijwmn
Transmission Control Protocol (TCP) was designed to provide reliable end-to-end delivery of
data over unreliable networks. In practice, most TCP deployments have been carefully designed in the
context of wired networks. Ignoring the properties of wireless and Ad-hoc Networks can lead to TCP
implementations with poor performance. In a wireless network, however packet losses occur more often
due to unreliable wireless links than due to congestion. When using TCP over wireless links, each packet
loss on the wireless link results in congestion control measures being invoked at the source. This causes
severe performance degradation. If there is any packet loss in wireless networks, then the reason for that
has to be found out. If there is congestion, then only congestion control mechanism has to be applied.
This work shows the performance of TCP with Adaptive Pacing (TCP-AP) and Link Random Early
Discard (LRED) as queuing model in multihop transmission when the source and destination nodes are
in mobile nature. The adaptive pacing technique seeks to improve spatial reuse. The LRED technique
seeks to react earlier to link overload. This paper consists of simulated environment results under
different network scenarios. This work proves that the combination of TCP-AP and LRED give much
better result than as the individual technique. Simulations are done with the use of NS-2.
The document provides an overview of Janet Abbate's book "Inventing the Internet" which explores the history of the development of the Internet from 1959 to 1994. The book examines the social and cultural factors influencing the Internet's evolution from ARPANET to a global network. It analyzes how the Internet was shaped by collaboration and conflict between various players including government, military, computer scientists, and businesses. The author traces the technological development of the Internet and links it to organizational, social, and cultural changes during that period.
IP resides at the network layer of the OSI model and provides logical addressing that allows systems on different logical networks to communicate. IP packets contain source and destination addresses as well as other fields. Transport protocols like UDP and TCP run on top of IP, with UDP being connectionless and used for real-time voice traffic in VoIP due to its simplicity and lower latency compared to TCP, which provides reliability but higher latency through mechanisms like acknowledgments and retransmissions. RTP runs on top of UDP to provide additional timestamping and sequencing information important for applications like voice calling.
This document describes a custom reliable file transfer protocol over UDP that is designed to perform better than TCP in lossy network conditions. It discusses the protocol design which uses sequence numbers and negative acknowledgements to provide reliability over UDP. The protocol is tested on a simulated network with varying packet loss rates and delays. Results show the protocol achieves higher throughput than TCP-based file transfer methods on lossy links.
Manoj Datt presented on the Sky X technology for improving performance over satellite links. The Sky X system uses Xpress Transport Protocol (XTP) and a Sky X Gateway to optimize data transfer speeds over satellites. TCP is not well suited for satellite conditions that involve long delays and packet loss. However, with Sky X Gateways on either side of the satellite link performing TCP to XTP conversion, performance can be increased up to 3 times for web usage and 10 to 100 times for file transfers, without any changes required to end clients and servers. The Sky X technology provides a reliable and efficient solution for satellite communication networks.
Manoj Datt presented on the Sky X technology for improving performance over satellite links. The Sky X system uses Xpress Transport Protocol (XTP) and a Sky X Gateway to optimize data transfer speeds over satellites. TCP is not well suited for satellite conditions that involve long delays and high bit errors. The Sky X Gateway replaces TCP with XTP for the satellite hop and back to TCP on the ground, improving speeds by 3-10x without any changes needed to end devices and applications. This Sky X architecture provides fully reliable, fast transmissions over satellites.
The transport layer provides end-to-end communication between processes on different machines. Two main transport protocols are TCP and UDP. TCP provides reliable, connection-oriented data transmission using acknowledgments and retransmissions. UDP provides simpler, connectionless transmission but without reliability. Both protocols use port numbers to identify processes and negotiate quality of service options during connection establishment.
White Paper: Accelerating File TransfersFileCatalyst
Check out our White Paper on Accelerating File Transfers
Increase File Transfer Speeds in Poorly-Performing Networks for an understanding of the issues associated with transferring files over the TCP/IP protocol (i.e. using FTP) and how to solve these problems with file transfer acceleration.
Efficient and Fair Bandwidth Allocation AQM Scheme for Wireless NetworksCSCJournals
Heterogeneous Wireless Networks are considered nowadays as one of the potential areas in research and development. The traffic management’s schemes that have been used at the fusion points between the different wireless networks are classical and conventional. This paper is focused on developing a novel scheme to overcome the problem of traffic congestion in the fusion point router interconnected the heterogeneous wireless networks. The paper proposed an EF-AQM algorithm which provides an efficient and fair allocation of bandwidth among different established flows. Finally, the proposed scheme developed, tested and validated through a set of experiments to demonstrate the relative merits and capabilities of a proposed scheme
The NaVeOl method is a patented data encoding technique that allows for absolute and relative operations on data sizes and formats that standard algorithms cannot handle. A special application of this method is NaVeOl Script, an encryption program that incorporates dynamic encryption and "iceberg coding" for security. Compared to other encryption methods like Blowfish and homomorphic encryption, NaVeOl Script is significantly faster while maintaining a high level of encryption complexity. The NaVeOl method and 8to7 technology more broadly cover non-standard algorithmic operations and data representation and have applications in data compression, encryption, and beyond.
This document discusses how blockchain can help enterprises by increasing trust, accountability, and transparency across business networks. It explains that blockchain enables new business models by leveraging partner ecosystems more efficiently. Enterprises need to develop blockchain strategies now to improve growth, react to competition, and increase security. The document provides advice on how to strategically identify opportunities and vulnerabilities when developing a blockchain strategy. It also outlines how IBM's blockchain services can help with industry expertise, open source leadership, demonstrated success, security, and individualized attention.
This document discusses how blockchain can help businesses by increasing trust, accountability, and transparency across business networks. It explains that blockchain enables new business models by leveraging partner ecosystems more efficiently. It argues that businesses need to develop blockchain strategies now to improve growth, react to competition, and increase security. The document provides advice on how to strategically identify opportunities and vulnerabilities when developing a blockchain plan. It highlights how IBM offers industry expertise, development platforms, and success enabling blockchain for business growth while ensuring security and regulatory compliance.
This document discusses how IBM's QRadar security intelligence platform can enable service providers to extend security capabilities to customers through multi-tenancy and software-as-a-service (SaaS) delivery models. It describes QRadar's multi-tenant capabilities that allow a single deployment to securely support multiple customer domains. It also introduces the QRadar Master Console, which provides centralized monitoring and management across multiple QRadar systems. Finally, it discusses how service providers can deploy QRadar in the cloud through IBM Security Intelligence on Cloud to minimize costs and offer an operating expense model.
Bluemix is IBM's cloud platform that allows developers to build, deploy, and manage applications. It provides flexible compute options including containers, virtual machines, and Cloud Foundry applications. Bluemix is underpinned by open technologies like Cloud Foundry, Docker, and OpenStack. It offers deployment options in IBM's public cloud, in private dedicated environments, and locally on-premises. Bluemix provides a catalog of services, robust DevOps tooling, integration capabilities, and runtimes to build and extend applications.
IBM Spectrum Storage was introduced in February 2015 as a family of optimized software-defined storage solutions. It provides unified analytics-driven management for any storage, advanced data placement to maximize performance and efficiency, and deployment flexibility. Over the last 6 months, IBM Spectrum Storage has added several new capabilities including management, protection, archiving, virtualization, acceleration, and scaling solutions. It offers comprehensive data protection with dedupe that can reduce backup infrastructure costs by up to 53%.
Skybuffer SAM4U tool for SAP license adoptionTatiana Kojar
Manage and optimize your license adoption and consumption with SAM4U, an SAP free customer software asset management tool.
SAM4U, an SAP complimentary software asset management tool for customers, delivers a detailed and well-structured overview of license inventory and usage with a user-friendly interface. We offer a hosted, cost-effective, and performance-optimized SAM4U setup in the Skybuffer Cloud environment. You retain ownership of the system and data, while we manage the ABAP 7.58 infrastructure, ensuring fixed Total Cost of Ownership (TCO) and exceptional services through the SAP Fiori interface.
In the rapidly evolving landscape of technologies, XML continues to play a vital role in structuring, storing, and transporting data across diverse systems. The recent advancements in artificial intelligence (AI) present new methodologies for enhancing XML development workflows, introducing efficiency, automation, and intelligent capabilities. This presentation will outline the scope and perspective of utilizing AI in XML development. The potential benefits and the possible pitfalls will be highlighted, providing a balanced view of the subject.
We will explore the capabilities of AI in understanding XML markup languages and autonomously creating structured XML content. Additionally, we will examine the capacity of AI to enrich plain text with appropriate XML markup. Practical examples and methodological guidelines will be provided to elucidate how AI can be effectively prompted to interpret and generate accurate XML markup.
Further emphasis will be placed on the role of AI in developing XSLT, or schemas such as XSD and Schematron. We will address the techniques and strategies adopted to create prompts for generating code, explaining code, or refactoring the code, and the results achieved.
The discussion will extend to how AI can be used to transform XML content. In particular, the focus will be on the use of AI XPath extension functions in XSLT, Schematron, Schematron Quick Fixes, or for XML content refactoring.
The presentation aims to deliver a comprehensive overview of AI usage in XML development, providing attendees with the necessary knowledge to make informed decisions. Whether you’re at the early stages of adopting AI or considering integrating it in advanced XML development, this presentation will cover all levels of expertise.
By highlighting the potential advantages and challenges of integrating AI with XML development tools and languages, the presentation seeks to inspire thoughtful conversation around the future of XML development. We’ll not only delve into the technical aspects of AI-powered XML development but also discuss practical implications and possible future directions.
Let's Integrate MuleSoft RPA, COMPOSER, APM with AWS IDP along with Slackshyamraj55
Discover the seamless integration of RPA (Robotic Process Automation), COMPOSER, and APM with AWS IDP enhanced with Slack notifications. Explore how these technologies converge to streamline workflows, optimize performance, and ensure secure access, all while leveraging the power of AWS IDP and real-time communication via Slack notifications.
TrustArc Webinar - 2024 Global Privacy SurveyTrustArc
How does your privacy program stack up against your peers? What challenges are privacy teams tackling and prioritizing in 2024?
In the fifth annual Global Privacy Benchmarks Survey, we asked over 1,800 global privacy professionals and business executives to share their perspectives on the current state of privacy inside and outside of their organizations. This year’s report focused on emerging areas of importance for privacy and compliance professionals, including considerations and implications of Artificial Intelligence (AI) technologies, building brand trust, and different approaches for achieving higher privacy competence scores.
See how organizational priorities and strategic approaches to data security and privacy are evolving around the globe.
This webinar will review:
- The top 10 privacy insights from the fifth annual Global Privacy Benchmarks Survey
- The top challenges for privacy leaders, practitioners, and organizations in 2024
- Key themes to consider in developing and maintaining your privacy program
leewayhertz.com-AI in predictive maintenance Use cases technologies benefits ...alexjohnson7307
Predictive maintenance is a proactive approach that anticipates equipment failures before they happen. At the forefront of this innovative strategy is Artificial Intelligence (AI), which brings unprecedented precision and efficiency. AI in predictive maintenance is transforming industries by reducing downtime, minimizing costs, and enhancing productivity.
Your One-Stop Shop for Python Success: Top 10 US Python Development Providersakankshawande
Simplify your search for a reliable Python development partner! This list presents the top 10 trusted US providers offering comprehensive Python development services, ensuring your project's success from conception to completion.
How to Interpret Trends in the Kalyan Rajdhani Mix Chart.pdfChart Kalyan
A Mix Chart displays historical data of numbers in a graphical or tabular form. The Kalyan Rajdhani Mix Chart specifically shows the results of a sequence of numbers over different periods.
Fueling AI with Great Data with Airbyte WebinarZilliz
This talk will focus on how to collect data from a variety of sources, leveraging this data for RAG and other GenAI use cases, and finally charting your course to productionalization.
Taking AI to the Next Level in Manufacturing.pdfssuserfac0301
Read Taking AI to the Next Level in Manufacturing to gain insights on AI adoption in the manufacturing industry, such as:
1. How quickly AI is being implemented in manufacturing.
2. Which barriers stand in the way of AI adoption.
3. How data quality and governance form the backbone of AI.
4. Organizational processes and structures that may inhibit effective AI adoption.
6. Ideas and approaches to help build your organization's AI strategy.
Skybuffer AI: Advanced Conversational and Generative AI Solution on SAP Busin...Tatiana Kojar
Skybuffer AI, built on the robust SAP Business Technology Platform (SAP BTP), is the latest and most advanced version of our AI development, reaffirming our commitment to delivering top-tier AI solutions. Skybuffer AI harnesses all the innovative capabilities of the SAP BTP in the AI domain, from Conversational AI to cutting-edge Generative AI and Retrieval-Augmented Generation (RAG). It also helps SAP customers safeguard their investments into SAP Conversational AI and ensure a seamless, one-click transition to SAP Business AI.
With Skybuffer AI, various AI models can be integrated into a single communication channel such as Microsoft Teams. This integration empowers business users with insights drawn from SAP backend systems, enterprise documents, and the expansive knowledge of Generative AI. And the best part of it is that it is all managed through our intuitive no-code Action Server interface, requiring no extensive coding knowledge and making the advanced AI accessible to more users.
HCL Notes and Domino License Cost Reduction in the World of DLAUpanagenda
Webinar Recording: https://www.panagenda.com/webinars/hcl-notes-and-domino-license-cost-reduction-in-the-world-of-dlau/
The introduction of DLAU and the CCB & CCX licensing model caused quite a stir in the HCL community. As a Notes and Domino customer, you may have faced challenges with unexpected user counts and license costs. You probably have questions on how this new licensing approach works and how to benefit from it. Most importantly, you likely have budget constraints and want to save money where possible. Don’t worry, we can help with all of this!
We’ll show you how to fix common misconfigurations that cause higher-than-expected user counts, and how to identify accounts which you can deactivate to save money. There are also frequent patterns that can cause unnecessary cost, like using a person document instead of a mail-in for shared mailboxes. We’ll provide examples and solutions for those as well. And naturally we’ll explain the new licensing model.
Join HCL Ambassador Marc Thomas in this webinar with a special guest appearance from Franz Walder. It will give you the tools and know-how to stay on top of what is going on with Domino licensing. You will be able lower your cost through an optimized configuration and keep it low going forward.
These topics will be covered
- Reducing license cost by finding and fixing misconfigurations and superfluous accounts
- How do CCB and CCX licenses really work?
- Understanding the DLAU tool and how to best utilize it
- Tips for common problem areas, like team mailboxes, functional/test users, etc
- Practical examples and best practices to implement right away
Trusted Execution Environment for Decentralized Process MiningLucaBarbaro3
Presentation of the paper "Trusted Execution Environment for Decentralized Process Mining" given during the CAiSE 2024 Conference in Cyprus on June 7, 2024.
Salesforce Integration for Bonterra Impact Management (fka Social Solutions A...Jeffrey Haguewood
Sidekick Solutions uses Bonterra Impact Management (fka Social Solutions Apricot) and automation solutions to integrate data for business workflows.
We believe integration and automation are essential to user experience and the promise of efficient work through technology. Automation is the critical ingredient to realizing that full vision. We develop integration products and services for Bonterra Case Management software to support the deployment of automations for a variety of use cases.
This video focuses on integration of Salesforce with Bonterra Impact Management.
Interested in deploying an integration with Salesforce for Bonterra Impact Management? Contact us at sales@sidekicksolutionsllc.com to discuss next steps.
Salesforce Integration for Bonterra Impact Management (fka Social Solutions A...
IBM & Aspera
1. Technical Whitepaper
IBM Software
Aspera FASP
high-speed transport
A critical technology comparison to alternative
TCP-based transfer technologies
Highlights
Challenges
The Transmission Control Protocol (TCP) provides reliable data
delivery under ideal conditions, but has an inherent throughput
bottleneck that becomes obvious, and severe, with increased packet loss
and latency found on long-distance WANs. Adding more bandwidth does
not change the effective throughput. File transfer speeds do not improve
and expensive bandwidth is underutilized.
Solutions
Unlike TCP, FASP throughput is independent of network delay and
robust to extreme packet loss. FASP transfer times are as fast as possible
(up to 1,000x standard FTP) and highly predictable, regardless of
network conditions. The maximum transfer speed is limited only by the
resources of the endpoint computers (typically disk throughput).
Benefits
• Maximum speed and reliability
• Extraordinary bandwidth control
• Built-in security
• Flexible and open architecture
Contents:
2 Introduction
3 High-speed TCP overview
5 UDP-based high-speed solutions
10 Aspera®
FASP®
solution
2. provide an alternative means to achieve apparently higher
throughputs, but at tremendous bandwidth cost. These
approaches retransmit significant, sometimes colossal amounts
of unneeded file data, redundant with data already in flight or
received, and thus take many times longer to transfer file data
than is necessary, and cause huge bandwidth cost. Specifically,
their throughput of useful bits excluding retransmitted data
packets – “goodput” – is very poor. These approaches
deceptively appear to improve network bandwidth utilization
by filling the pipe with waste, and transfer times are still slow!
For the narrow network conditions under which TCP
optimizations or simple blasters do achieve high “good data”
throughput, as network-centric protocols, they run up
against further soft bottlenecks in moving data in and out of
storage systems.
Transporting bulk data with maximum speed calls for an
end-to-end approach that fully utilizes available bandwidth
along the entire transfer path – from data source to data
destination – for transport of “good data” – data that is not in
flight or yet received. Accomplishing this goal across the great
range of network round-trip times, loss rates and bandwidth
capacities characteristic of the commodity Internet WAN
environments today requires a new and innovative approach to
bulk data movement, specifically, an approach that fully
decouples reliability and rate control. In its reliability
mechanism, the approach should retransmit only needed data,
for 100 percent good data throughput. In its rate control, for
universal deployment on shared Internet networks, the
approach should uphold the principles of bandwidth fairness,
and congestion avoidance in the presence of other transfers and
other network traffic, while providing the option to dedicate
bandwidth for high priority transfers when needed.
Aspera FASP is an innovative bulk data transport technology
built on these core principles that is intended to provide an
optimal alternative to traditional TCP-based transport
technologies for transferring files over public and private
IP networks. It is implemented at the application layer, as an
endpoint application protocol, avoiding any changes to
standard networking. FASP is designed to deliver 100 percent
bandwidth efficient transport of bulk data over any IP network
– independent of network delay and packet loss – providing the
ultimate high-performance next-generation approach to
moving bulk data.
IBM Software
2
Technical Whitepaper
Introduction
In this digital world, fast and reliable movement of digital
data, including massive sizes over global distances, is
becoming vital to business success across virtually every
industry. The Transmission Control Protocol (TCP) that has
traditionally been the engine of this data movement, however,
has inherent bottlenecks in performance (Figure 1), especially
for networks with high, round-trip time (RTT) and packet
loss, and most pronounced on high-bandwidth networks. It is
well understood that these inherent “soft” bottlenecks are
caused by TCP’s Additive-Increase-Multiplicative-Decrease
(AIMD) congestion avoidance algorithm, which slowly probes
the available bandwidth of the network, increasing the
transmission rate until packet loss is detected and then
exponentially reducing the transmission rate. However, it is
less understood that other sources of packet loss, such as
losses due to the physical network media, not associated with
network congestion equally reduce the transmission rate. In
fact, TCP AIMD itself creates losses, and equally contributes
to the bottleneck. In ramping up the transmission rate until
loss occurs, AIMD inherently overdrives the available
bandwidth. In some cases, this self-induced loss actually
surpasses loss from other causes (e.g., physical media or bursts
of cross traffic) and turns a loss-free communication “channel”
to an unreliable “channel” with an unpredictable loss ratio.
The loss-based congestion control in TCP AIMD has a very
detrimental impact on throughput: Every packet loss leads to
retransmission, and stalls the delivery of data to the receiving
application until retransmission occurs. This can slow the
performance of any network application, but is fundamentally
flawed for reliable transmission of large “bulk” data, for
example file transfer, which does not require in-order (byte
stream) delivery.
This coupling of reliability (retransmission) to congestion
control in TCP creates a severe artificial throughput penalty
for file transport, as evident by the poor performance of
traditional file transfer protocols built on TCP such as
FTP, HTTP, CIFS, and NFS over wide area networks.
Optimizations for these protocols such as TCP acceleration
applied through hardware devices or alternative TCP improve
file transfer throughput to some degree when round-trip
times and packet loss rates are modest, but the gains diminish
significantly at global distances. Furthermore, as we will see
later in this paper, parallel TCP or UDP blasting technologies
3. However, the improvement diminishes rapidly in wide area
networks, where packet losses due to physical media error or
buffer overflow by cross traffic bursts become non-negligible.
A single packet loss in these networks will cause the TCP
sending window to reduce severely, while multiple losses will
have a catastrophic effect on data throughput. More than one
packet loss per window typically results in a transmission timeout
and the resulting bandwidth-delay-product pipeline from sender
to receiver drains and data throughput drops to zero.The sender
essentially has to re-slow-start data transmission.
In contrast, in Aspera FASP the transmission rate is not
coupled to loss events. Lost data is retransmitted at a rate
corresponding to the end-to-end desired bandwidth. The
retransmission achieves virtually ideal bandwidth efficiency –
no data is transmitted in duplicate and the total target capacity
is fully utilized.
As shown in Figure 2, the throughput of FAST TCP, one such
commercial version of high-speed TCP (which include such
variations as CUBIC, H-TCP, BIC, etc.) on a network of
one percent packet loss improves the throughput over standard
TCP Reno on low latency networks, but the improvement falls
off rapidly at higher round-trip times typical of cross-country
and intercontinental links. The FASP throughput in contrast
has no degradation with increasing network delay and achieves
up to 100 percent efficient transmission and an effective file
transfer throughput at over 95 percent of the bandwidth
capacity. Similarly, as packet loss increases (e.g., at five percent
loss or more) the FASP throughput decreases only by the same
amount. At higher loss rates the accelerated TCP throughput
approximates Reno.
Figure 2: File transfer throughput for 1 GB file comparing Reno TCP, a
commercially available high-speed TCP, UDT, and Aspera FASP in a link
with medium packet loss (one percent). Note that while the accelerated
TCP improves Reno throughput on lower latency networks, the throughput
improvement falls off rapidly at higher round-trip times typical of cross-
country and intercontinental links. The FASP throughput in contrast has no
degradation with increasing delay. Similarly, as the packet loss rate
increases (e.g., at five percent loss) the FASP throughput decreases only by
about the same amount, while high-speed TCP is no better than Reno.
In this paper we describe the alternative approaches to
“accelerating” file-based transfers – both commercial and
academic – in terms of bandwidth utilization, network
efficiency, and transfer time, and compare their performance
and actual bandwidth cost to Aspera FASP.
Figure 1: The bar graph shows the maximum throughput achievable under
various packet loss and network latency conditions on an OC-3 (155 Mbps)
link for file transfer technologies that use TCP (shown in yellow). The
throughput has a hard theoretical limit that depends only on the network
RTT and the packet loss. Note that adding more bandwidth does not
change the effective throughput. File transfer speeds do not improve and
expensive bandwidth is underutilized.
High-speed TCP overview
In recent years, a number of new high-speed versions of the
TCP protocol and TCP acceleration appliances implementing
these variations have been developed. High-speed TCP
protocols recognize the fundamental flaw of AIMD and
revamp this window-based congestion control algorithm to
reduce the artificial bottleneck caused by it, and improve the
long-term average throughput. The most advanced versions
of these protocols typically aim to improve the detection of
congestion through measuring richer signals such as network
queuing delay, rather than increasing throughput until a loss
event occurs. This helps to prevent TCP flows from creating
packet loss, and thus artificially entering congestion
avoidance, and improves the long-term throughput in nearly
loss-free networks.
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Standard and High-Speed TCP’s reaction to packet loss forces
the sender to not only reduce its sending window, leading to
an erratic transfer speed, but also pre-empts new packets in
the sending window with retransmitted packets to maintain
TCPs in-order delivery guarantee. This transmission of new
and retransmitted packets in the same TCP sending window
entangles the underperforming TCP congestion control with
TCP reliability control that guarantees transfer integrity, and
unnecessarily handicaps transfer throughput for applications
that do not require in-order delivery, such as bulk data.
TCP reliability guarantees that no data is lost (the lost packets
will be detected by the receiver and retransmitted by the
sender afterwards), and received data is delivered, in-order, to
the application. In order to fulfill these two guarantees, TCP
not only retransmits the lost packets, but also stalls the
earlier-arriving, out-of-order packets (stored temporarily in
the kernel memory) until the missing packet arrives, and the
received data can be delivered to the application layer, in-
order. Given the requirement that the receiver must continue
storing incoming packets in RAM until the missing data is
received, retransmission is urgent and first priority, and the
sending of new data must be slowed in concert. Specifically, on
every packet loss event, new packets have to slow down
(typically the sending window freezes until lost packets are
retransmitted to receiver and acknowledged), waiting for
retransmitted packets to fill “holes” in the byte stream at
receiving end. In essence the reliability and flow control (or
congestion control) in TCP are by design, thoroughly coupled.
Although this type of mechanism provides TCP with a strict
in-order byte stream delivery required by many applications,
it becomes devastating to applications that naturally do not
require strict byte order, such as file transfer, and thus
introduces a hidden artificial bottleneck to these applications,
limiting their corresponding data throughput.
To make it crystal clear, we can explore a simple example to
calculate the throughput loss due to a single non-congestion-
related packet loss in a High-Speed TCP with a window
reduction of one-eighth on each loss. For a Gigabit network
with one percent packet loss ratio and 100 ms round-trip
delay, every single packet loss causes the rate to reduce by
one-eighth (compared with one half in TCP Reno) and it will
take 1 Gbps÷8(bits/byte)÷1024(bytes/packet)×100 ms× 0.125
(drop ratio/loss)×100ms ≈ 152.6 seconds for the sender to
recover the original sending speed (1 Gbps) before the packet
loss event. During this recovery period, High-speed TCP
loses about 152.6s×1 Gbps×0.125/2 ≈ 8.9 GB throughput
because of a single loss event! In the real wide area network,
the actual value will be even larger since RTT can be larger
due to network queuing, physical media access, scheduling
and recovery, etc. Thus it typically takes longer than 152.6
seconds for the sender to recover. Multiple consecutive packet
losses will be a catastrophe. A quote from Internet
Engineering Task Force (IETF) bluntly puts the effect in this
way:
“Expanding the window size to match the capacity of an LFN
[long fat network] results in a corresponding increase of the
probability of more than one packet per window being
dropped. This could have a devastating effect upon the
throughput of TCP over an LFN. In addition, since the
publication of RFC 1323, congestion control mechanisms
based upon some form of random dropping have been
introduced into gateways, and randomly spaced packet drops
have become common; this increases the probability of
dropping more than one packet per window.”1
We note that this rate slowdown or throughput loss is
sometimes indeed necessary for byte-stream applications
where strict in-order delivery is a must. Otherwise, RAM has
to accommodate at least 1 Gbps×100 ms×0.125 ≈ 1.5 MB
extra data just to wait for a single lost packet of each TCP
connection for at least one RTT in our earlier example.
However, this slowdown becomes unnecessary for file transfer
applications because out-of-order data can be written to disk
without waiting for this lost packet, which can be
retransmitted any time at the speed that precisely matches the
available bandwidth inside the network, discovered by an
advanced rate control mechanism.
Indeed TCP by itself will not be able to decouple reliability
and congestion control and thus will not remove this artificial
bottleneck unless the purposes of TCP – providing reliable,
byte-stream delivery – are redefined by the IETF.2
The
traditional reliance upon a single transmission control
protocol for both reliable streaming and non-streaming
applications has been proven in practice to be suboptimal for
both domains.
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Figure 3: The bar group shows the throughput achieved under various
packet loss and network latency (WAN) conditions on a 300 Mbps link for
RocketStream. Bars with zero height represent failures of establishing
connection between sender and receiver, which is not uncommon to
RocketStream when either RTT or packet loss ratio is large.
We selected one of the most advanced retransmission
(NACK-based) UDP transport solutions, “UDT”,
re-packaged by commercial vendors, to demonstrate these
problems. Specifically, they include:
• Poor congestion avoidance. The dynamic “AIMD” algorithm
(D-AIMD) employed in UDT behaves similarly to AIMD,
but with a decreasing additive increase (AI) parameter that
scales back the pace of rate increase, as the transmission
rate increases. This approach fails to recognize the
aforementioned key issues of TCP – the entanglement of
reliability and rate control – and instead makes the
assumption that tuning one parameter can solve the
underperformance of AIMD and even TCP. Indeed, a
specifically tuned D-AIMD outperforms TCP in one
scenario, but immediately underperforms TCP in another.
Thus, in many typical wide area networks, the performance
of UDT is actually worse than TCP.
UDP-based high-speed solutions
The reliability provided by TCP reduces network throughput,
increases average delay and worsens delay jitter. Efforts to
decouple reliability from congestion avoidance have been
made for years. Due to the complexity of changing TCP itself,
in recent years academic and industry practices have pursued
application-level protocols that feature separable rate and
reliability controls. These approaches use UDP in the
transport layer as an alternative to TCP and implement
reliability at the application layer. Most such approaches are
UDP blasters – they move data reliably with UDP, employing
some means to retransmit lost data - but without meaningful
consideration of the available bandwidth, and risk network
collapse, not to mention collapse of their own throughput.
Figure 3 shows the throughput of Rocketstream, a commercial
UDP data blaster, when run over a 300 Mbps link with typical
WAN conditions (increasing RTT and packet loss).
UDP solutions, including open source implementations
Tsunami and UDT (used by products such as Signiant, File
Catalyst, and Sterling Commerce®
), have attempted to
strengthen congestion control in UDP blasters through
simplistic algorithms that reduce transmission rate in the face
of packet loss. While the back off can be “tuned” to achieve
reasonable performance for specific network pathways on a
case-by-case basis, meaning single combinations of bandwidth,
round-trip delay, packet loss and number of flows, the design
is inherently unable to adapt to the range of network RTT
and packet loss conditions and flow concurrency in any
real-world Internet network. Consequently, these approaches
either underutilize available bandwidth, or apparently “fill the
pipe”, but in the process overdrive the network with
redundant data transmission – as much as 50 percent
redundant data under typical network conditions - that wastes
bandwidth in the first order, and leads to collapse of the
effective file transfer throughput (“goodput”) in the second
order. Finally, in the process these approaches can leave the
network unusable by other traffic as their overdrive creates
packet loss for other TCP applications and stalls their
effective throughput.
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• Aggressive sending and flawed retransmission in UDT leads
to lower efficiency of valuable bandwidth and often forces
customers to purchase more bandwidth unnecessarily. (The
very solution that is intended to better utilize expensive
bandwidth actually wastes it.) The large difference between
sending rate, receiving rate, and effective file transfer rate in
some experiments (Figures 6 and 7) exposes the significant
data drops at the router and the receiver primarily due to
overly aggressive data injection and the flawed retransmission
mechanism of UDT. Measured efficiency (“goodput”) drops
below 20 percent in some typical wide area networks. That
means a 100 percent fully utilized network by UDT uses 80
percent of the bandwidth capacity transmitting redundant
(duplicate) data to receiver or transmitting useful data to an
overflowed buffer (overdriven by UDT itself).
The “benefit” and “cost” of using UDT to a regular user can
be quantified for an accurate comparison. “Benefit” can be
measured by the efficient use of bandwidth for transferring
needed data (goodput) translating directly into speedy transfer
times, while the “cost” can be abstracted as the effort of
transferring one needed data packet, defined as how many
duplicated copies are transferred to get one needed packet
successfully delivered to the application layer at another
network end. This cost also implies the induced costs to other
transfers, reflected by their loss of fair bandwidth share (Figure
4) and thus their degraded throughputs. Specifically, as already
partially reflected in Figure 5, UDT has lower effective
transfer throughput (resulting in slow transfers) over a wide
range of WAN conditions, and thus brings little benefit to
users. And, the associated bandwidth cost due to overdrive and
redundant retransmission dramatically affects other
workflows.
Figure 5 shows the overall cost of transmitting one packet by a
single UDT transfer on a T3 (45 Mbps) link under different
RTTs and packet loss ratios. For most typical wide area
networks, one packet transfer needs eight to ten
retransmissions. In another words, in order to transfer a
1-gigabyte file, the UDT sender dumps nine to eleven
gigabytes into the network in the end. The transfer takes 9 –
11 times longer than necessary, and also causes large packet
loss to other flows.
• UDT’s aggressive data sending mechanism causes dramatic
rate oscillation and packet loss, not only undermining its own
throughput, but also jeopardizing other traffic and degrading
overall network performance. In a typical wide area network
where a regular TCP flow (e.g., a HTTP session of a web
client) shares bandwidth with a UDT transfer, the TCP flow
can potentially experience denial-of-service due to
aggressiveness of the UDT flow (Figure 4). Indeed, the
possibility of extreme TCP unfriendliness is theoretically
studied in the original UDT paper, “Optimizing UDP-based
Protocol Implementations, 2005”, and the authors propose a
specific condition that must be satisfied to avoid this extreme
unfriendliness.2
In reality, for very typical wide area networks
(e.g., WAN with 100 ms RTT and 0.1 percent plr), the
condition cannot be satisfied and thus this extreme TCP
unfriendliness will be inevitable. That means in order to use a
UDT-based data movement solution, a regular customer will
likely need to invest more time and money on some type of
QoS companion to guarantee UDT will not damage the
operation of the whole network ecosystem (e.g., web, email,
VOIP, network management).
Figure 4: File transfer throughput of a single UDT transfer on a typical T3 link
with zero percent packet loss ratio and 50 ms RTTs, and the effect of UDT
transfer on a regular TCP flow. The TCP flow is not “visible” for most of its
duration until the UDT flow terminates.
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The cost in Figure 5 is caused by the overly aggressive injection rate of the UDT sender and duplicate retransmissions dropped by
the UDT receiver. To be more specific, we can define sending cost to reflect the loss due to an overly aggressive injection by the
sender and thus packet drops at router, and the receiving cost to reflect duplicate retransmissions dropped at receiver.
More accurately, the sending cost is
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Figure 6: The bar graph shows the sending rates, receiving rates, and effective rates of a single UDT transfer under different RTTs and packet loss ratios on a T3
link. Note that the large difference between sending and receiving rate implies large packet loss on the intervening network path, and the large difference in the
receiving and effective rate implies a large number of duplicate retransmissions.
Figure 5: The bar graph shows the retransmissions of a single UDT transfer under different RTTs and packet loss ratios. The height of each bar, the “transmission
cost,” is the quantity of retransmitted data in units of gigabytes when a 1 GB file is transferred. Bars with zero height represent failures to establish a connection
between sender and receiver, which is not uncommon to UDT when either RTT or packet loss ratio is large. Note that up to nine times the original file size is sent
and in wasteful retransmissions.
and the receiving cost is
8. IBM Software
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Note that the higher the sending cost, the more packets are
dropped at the router, while the higher the receiving cost, the
more packets are dropped at receiver. Figure 7 shows the
sending rates, receiving rates, and effective rates of a single
UDT transfer under different RTTs and packet loss ratios on
a T3 link. The rates ratios (sending rate to receiving rate and
receiving rate to effective rate) will be the defined costs above.
We observe that sending rates are persistently higher than
receiving rates, which are again persistently higher than
effective rate in all network configurations. These costs
drive the network to an operational point where network
utilization (defined as throughput divided by bandwidth)
is close to one, but the network efficiency (defined as
goodput divided by bandwidth) is as low as 15 percent.
Consequently, any given file transfer is over six times
slower than it should be.
To be crystal clear, we can verify the above costs through a
simple file transfer example under different wide area
networks by answering the following performance-
related questions:
• How many bytes are to be sent?
• How many bytes are actually sent?
• How many bytes are actually received?
• How long has the transfer taken?
• What is the effective file transfer speed?
The answers are organized in Table 1, and compared to
Aspera FASP for the same conditions.
Figure 7: The sending, receiving, and effective receiving rates of a single
UDT transfer on a T3 Link with zero percent, one percent and five percent
packet loss ratios and 100 ms, 200 ms, 200 ms RTTs. The gap between
sending and receiving rates implies large amount of data loss at router,
while the gap between receiving and effective receiving rates reflects the
large number of drops of duplicate retransmissions at the UDT receiver.
(a):UDT transfer on a T3 Link with zero percent plr and 100 ms RTT
(b):UDT transfer on a T3 Link with one percent plr and 100 ms RTT
(c):UDT transfer on a T3 Link with five percent plr and 200 ms RTT
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Bandwidth
(Mbps)
RTT
(ms)
plr
(%)
How
much to
be sent
(MB)
How much
needs to be
sent (actual
data +
inevitable loss
by media, MB)
How
much
data
actually
sent
(MB)
Sending
cost
(Sender’s
Overhead,
%)
How
much
data
actually
received?
(MB)
Receiving
Cost
(Receiver’s
Overhead,
%)
How long
does it
take? (s)
Effective
file
transfer
speed
(Mbps)
Observed
Network
Utilization
Network
Efficiency
(Effective
Utilization,
%)
45 0 0 953.7 953.7 9093.2 314.1% 2195.8 130.2% 625.0 12.8 66.2% 28.4%
45 100 1 953.7 963.2 5941.8 234.9% 1774.0 86.0% 618.0 12.9 54.1% 28.8%
45 400 5 953.7 1001.4 3764.1 150.6% 1501.8 57.5% 830.0 9.6 34.1% 21.4%
45 800 5 953.7 1001.4 3549.9 152.9% 1403.9 47.2% 1296.0 6.2 20.4% 13.7%
100 100 1 953.7 963.2 1413.0 14.0% 1239.8 30.0% 239.0 33.5 44.0% 33.5%
100 200 5 953.7 1001.4 2631.2 19.6% 2200.1 130.7% 571.8 14.0 32.6% 14.0%
300 100 1 953.7 963.2 1060.0 2.1% 1038.4 8.9% 232.0 34.5 12.6% 11.5%
300 200 1 953.7 963.2 1083.0 2.3% 1059.1 11.1% 273.0 29.3 11.0% 9.8%
500 200 1 953.7 963.2 1068.9 1.7% 1051.5 10.3% 252.0 31.8 7.1% 6.4%
500 200 5 953.7 1001.4 1660.9 5.3% 1576.7 65.3% 539.1 14.8 5.0% 3.0%
Table 1: UDT file transfer over typical WANs – high-bandwidth cost and slow transfer rate
Bandwidth
(Mbps)
RTT
(ms)
plr
(%)
How
much
data to
be sent?
(MB)
How much
needs to be sent
(actual data +
inevitable loss
by media, MB)
How
much
data
actually
sent?
(MB)
Sending
cost
(Sender’s
Overhead,
%)
How
much
data
actually
received?
(MB)
Receiving
Cost
(Receiver’s
Overhead,
%)
How long
does it
take? (s)
Effective
file
transfer
speed
(Mbps)
Observed
Network
Utilization
(by
receiver)
Network
Efficiency
(Effective
Utilization,
%)
45 0 0 953.7 953.7 953.7 0.0% 953.7 0.0% 185.4 43.1 98.5 95.9%
45 100 1 953.7 963.2 963.3 1.0% 953.7 0.0% 187.8 42.6 97.1 94.6%
45 400 5 953.7 1001.4 1002.1 5.0% 954.3 0.1% 197.0 40.6 92.1 90.3%
45 800 5 953.7 1001.4 1003.5 5.1% 955.2 0.2% 197.0 40.6 91.6 90.3%
100 100 1 953.7 963.2 963.3 1.0% 953.8 0.0% 85.0 94.1 96.3 94.1%
100 200 5 953.7 1001.4 1002.4 5.0% 954.5 0.1% 88.9 90.0 91.9 90.0%
300 100 1 953.7 963.2 964.0 1.0% 954.4 0.1% 29.3 273.4 92.6 91.1%
300 200 1 953.7 963.2 964.7 1.0% 955.1 0.1% 29.2 274.3 91.9 91.4%
500 200 1 9536.7 9632.1 9635.0 1.0% 9539.0 0.0% 181.6 440.6 90.6 88.1%
500 200 5 9536.7 10013.6 10018.5 5.0% 9541.2 0.0% 186.9 428.0 88.0 85.6%
Table 2: Aspera FASP file transfer over typical WANs – near zero bandwidth cost and fast transfer rate
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Aspera FASP solution
Aspera FASP fills the gap left by TCP in providing reliable
transport for applications that do not require byte-stream
delivery and completely separates reliability and rate control.
It uses standard UDP in the transport layer and achieves
decoupled congestion and reliability control in the application
layer through a theoretically optimal approach that
retransmits precisely the real packet loss on the channel.
Due to the decoupling of the rate control and reliability, new
packets need not slow down for the retransferring of lost
packets as in TCP-based byte streaming applications. Data
that is lost in transmission is retransmitted at a rate that
matches the available bandwidth inside the end-to-end path,
or a configured target rate, with zero duplicate
retransmissions for zero receiving cost.
The available bandwidth inside the path is discovered by a
delay-based rate control mechanism, for near zero sending
cost. Specifically, FASP adaptive rate control uses measured
queuing delay as the primary indication of network (or
disk-based) congestion with the aim of maintaining a small,
stable amount of “queuing” in the network; a transfer rate
adjusts up as the measured queuing falls below the target
(indicating that some bandwidth is unused and the transfer
should speed up), and adjusts down as the queuing increases
above the target (indicating that the bandwidth is fully utilized
and congestion is eminent). By sending periodically probing
packets into the network, FASP is able to obtain a more
accurate and timely measurement of queuing delay along the
transfer path. When detecting rising queuing delay, a FASP
session reduces its transfer rate, proportional to the difference
between the target queuing and the current queuing,
therefore avoiding overdriving the network. When network
congestion settles down, the FASP session quickly increases
according to a proportion of the target queuing and thus
ramps up again to fully utilize nearly 100 percent of the
available network capacity.
The direct consequences of UDT file transfer performance
shown in Table 1 are that useful data does not get through the
network at all, or that it does so at the price of network
efficiency (shown in the eighth column of Table 1), which not
only compounds the poor performance, but also causes a
denial-of-service for other network traffic by saturating the
bandwidth. Note that creating parallel transfers for higher
sending rates and network utilization as employed in some
UDT and TCP solutions only aggravates bandwidth waste
and forces customers to invest in more bandwidth
prematurely. The consequent improvement in network
utilization and data throughput is little, but the resulting cost
(Figure 8) is dramatically increased. Retransmission is
increased by another 40 percent with two UDT sessions.
For the same example (Table 1), UDT dumps as much as
13 GB to 15 GB data to network in order to successfully
deliver a less-than-1 GB file. Solutions using parallel TCP
or UDT transfers have similar or even worse performance as
shown in Figure 8.
Figure 8: Bandwidth cost of parallel UDT streams. The graph shows the
retransmission costs of two parallel UDT sessions for a single 1 GB file
transfer under different RTTs and packet loss ratios in a T3 network. The
height of each bar, referred to as transmission cost, represents the amount
of retransmission in units of gigabytes when the 1 GB file is transferred.
Bars with zero height represent failures of establishing connection between
sender and receiver, which is not uncommon to UDT when either RTT or
packet loss ratio is large. Note that almost 14 GB (14x) the size of the file is
retransmitted in the process.
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Figure 10: FASP shared link capacity with other FASP and standard TCP
traffic, achieving intra-protocol and inter-protocol fairness.
Figure 11: FASP uses available bandwidth when TCP is limited by network
condition, achieving complete fairness between FASP flows and with other
(TCP) traffic.
Unlike TCP’s rate control, the FASP adaptive rate control has
several major advantages: First, it uses network queuing delay
as the primary congestion signal and packet loss ratio as the
secondary signal, and thus obtains the precise estimation of
network congestion, not artificially slowing down over
networks with packet loss due to the media. Second, the
embedded quick response mechanism allows high-speed file
transfers to automatically slow down to allow for stable, high
throughput when there are many concurrent transfers, but
automatically ramp up to fully, efficiently utilize unused
bandwidth for more efficient delivery times. Third, the
advanced feedback control mechanism allows the FASP
session rate to more quickly converge to a stable equilibrium
rate that injects a target amount of queued bits into the buffer
at the congested router. Stable transmission speed and
queuing delay bring QoS experience to end users without
additional investment on QoS hardware or software. Delivery
time of data becomes predictable and data movement is
transparent to other applications sharing the same network.
Fourth, the full utilization of bandwidth, unlike NACK based
UDP blasters, introduces virtually no cost to the network and
network efficiency is kept around 100 percent.
Figure 9: The bar graph shows the throughput achieved under various
packet loss and network latency conditions on a 1 Gbps link for file transfer
technologies that use FASP innovative transfer technology. Bandwidth
efficiency does not degrade with network delay and packet loss.
In addition to efficiently utilizing available bandwidth,
the delay-based nature of FASP adaptive rate control
allows applications to build intentional prioritization in the
transport service. The built-in response to network queuing
provides a virtual “handle” to allow individual transfers to be
prioritized/de-prioritized to help meet application goals, such
as offering differentiated bandwidth priorities to concurrent
FASP transfers.
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About Aspera, an IBM Company
Aspera, an IBM company, is the creator of next-generation
transport technologies that move the world’s data at maximum
speed regardless of file size, transfer distance and network
conditions. Based on its patented, Emmy®
award-winning
FASP®
protocol, Aspera software fully utilizes existing
infrastructures to deliver the fastest, most predictable file-
transfer experience. Aspera’s core technology delivers
unprecedented control over bandwidth, complete security and
uncompromising reliability. Organizations across a variety of
industries on six continents rely on Aspera software for the
business-critical transport of their digital assets.
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For more information on IBM Aspera solutions, please visit
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By removing artificial bottlenecks in network transport and
freeing up full link bandwidth to end users, FASP transfers
sometimes reveal newly emerging bottleneck points in Disk
IO, file systems, and CPU scheduling, etc., which inevitably
create new hurdles as the transmission rate is pushed to the
full line speed especially in multi-Gigabit networks. The
FASP adaptive rate control has been extended to include disk
flow control to avoid data loss in fast file transfer writing to a
relatively slow storage pathway. A similar delay-based model
(patent-pending) was developed for the disk buffer. Due to the
different time scales of network and disk dynamics, a two-
time-scale design was employed to accommodate both
bandwidth and disk speed changes. At a fine-grained, fast time
scale, a local feedback mechanism is introduced at the receiver
end to accommodate periodic disk slowdown due to operating
system scheduling as an example, while at a coarse-grained,
slow time scale, a unified delay-based congestion avoidance is
implemented for both bandwidth control and disk control,
enabling FASP transfers to simultaneously adapt to available
network bandwidth as well as disk speed.
File system bottlenecks manifest in a variety of aspects.
Indeed, many customers experience dramatically decreased
speed when transferring sets of small files compared with
transferring a single file of the same size. Using a novel file
streamlining technique, FASP removes the artificial
bottleneck caused by file systems and achieves the same ideal
efficiency for transfers of large numbers of small files.
For example, one thousand 2 MB files can be transmitted
from the US to New Zealand with an effective transfer speed
of 155 Mbps, filling an entire OC-3.
As a result, FASP eliminates the fundamental bottlenecks of
TCP- or UDP-based file transfer technologies such as FTP
and UDT, and dramatically speeds up transfers over public
and private IP networks. FASP removes the artificial
bottlenecks caused by imperfect congestion control
algorithms, packet losses (by physical media, cross-traffic
burst, or coarse protocols themselves), and the coupling
between reliability and congestion control. In addition, FASP
innovation is eliminating emerging bottlenecks from disk IO,
file system, CPU scheduling, etc. and achieves full line speed
on even the longest, fastest wide area networks. The result, we
believe, is a next-generation high-performance transport
protocol that fills the growing gap left by TCP for the
transport of large, file-based data at distance over commodity
networks, and thus makes possible the massive everyday
movement of digital data around the world.
Technical Whitepaper