Experiments for a Collaborative Haptic Virtual Reality

Classical Transmission Control Protocol (TCP) designs have never considered the identity of the competing transport protocol as useful information to TCP sources in congestion control mechanisms. When competing against a TCP flow on a bottleneck link, a User Datagram Protocol (UDP) flow can unfairly occupy the entire link bandwidth and suffocate all TCP flows on the link. If it were possible for a TCP source to know the type of transport protocol that deprives it of link access, perhaps it would be better for the TCP source to react in a way which prevents total starvation. In this paper, we use coefficient of variation and power spectral density of throughput traces to identify the presence of UDP transport protocols that compete against TCP flows on bottleneck links. Our results show clear traits that differentiate the presence of competing UDP flows from TCP flows independent of round-trip times variations. Signatures that we identified include an increase in coefficient of variation whenever a competing UDP flow joins the bottleneck link for the first time, noisy spectral density representation of a TCP flow when competing against a UDP flow in the bottleneck link, and a dominant frequency with outstanding power in the presence of TCP competition only. In addition, the results show that signatures for congestion caused by competing UDP flows are different from signatures due to congestion caused by competing TCP flows regardless of their round-trip times. The results in this paper present the first steps towards development of more ’intelligent’ congestion control algorithms with added capability of knowing the identity of aggressor protocols against TCP, and subsequently using this additional information for rate control.

In this paper, we develop a model-based frame scheduling scheme, called MFS, to enhance the capacity of IEEE 802.11-operated wireless local area networks (WLANs) for both transmission control protocol (TCP) and user datagram protocol (UDP) traffic. In MFS each node estimates the current network status by keeping track of the number of collisions it encounters between its two consecutive successful frame transmissions, and computes accordingly the current network utilization. The result is then used to determine a scheduling delay to be introduced before a node attempts to transmit its pending frame. MFS does not require any change in IEEE 802.11, but instead lays a thin layer between the LL and medium access control (MAC) layers. In order to accurately calculate the current utilization in WLANs, we develop an analytical model that characterizes data transmission activities in IEEE 802.11-operated WLANs with/without the request to send/clear to send (RTS/CTS) mechanism, and validate the model with ns-2 simulation. All the control overhead incurred in the physical and MAC layers, as well as system parameters specified in IEEE 802.11, are figured in. We conduct a comprehensive simulation study to evaluate MFS in perspective of the number of collisions, achievable throughput, intertransmission delay, and fairness in the cases of TCP and UDP traffic. The simulation results indicate that the performance improvement with respect to the protocol capacity in a WLAN of up to 300 nodes is 1) as high as 20% with the RTS/CTS and 70% without the RTS/CTS in the case of UDP traffic and 2) as high as 10% with the RTS/CTS and 40% without the RTS/CTS in the case of TCP traffic. Moreover, the intertransmission delay in MFS is smaller and exhibits less variation than that in IEEE 802.11; the fairness among wireless nodes in MFS is better than, or equal to, that in IEEE 802.11.

Applications operating over airborne networks are known to perform quite sub-optimally. Transmission Control Protocol (TCP) based applications often experience very low data throughput [1], while the User Datagram Protocol (UDP) based applications suffer high packet loss as well as out-of-order delivery [2]. This is due to the high Bit Error Rate (BER) nature of the wireless links and the high mobility of airborne nodes. Though TCP and UDP have performed well in wired networks, their original design did not take into consideration the characteristics seen in airborne networks thus resulting in their relatively poor performance. In this research, we implemented an OPNET model of the Reliable User Datagram Protocol (RUDP) as proposed in the IETF Internet-Draft in 1999 [3] with extensions to take into consideration the nature of airborne network characteristics. We then analyzed the performance and its applicability to airborne network scenarios. We also provide performance comparison of the enhanced RUDP against the TCP, UDP, and the standard RUDP counterparts.

<p>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.</p>

SCTP is a transport layer protocol mainly designed for transferring public switched telephone network signalling over IP networks. Some of the protocols that support data transmission are transmission control protocol (TCP), user datagram protocol (UDP), and stream control transmission protocol (SCTP). SCTP is a newly invented reliable, message-oriented transport protocol. It is similar to TCP and UDP with some additional features such as multi-streaming and multi-homing. Even though it is a new protocol with some advanced features, there is a security loophole issue in data transmission using SCTP. Hence, we proposed a new technique to transmit a secret message by hiding it in a secret matrix using the advanced features of SCTP. The simulation of our proposed method provides better performance and high security for secret message transmission against network attacks.

Generally, real-time applications based on the User Datagram Protocol (UDP) generate large volumes of data and are not sensitive to network congestion. In contrast, Transmission Control Protocol (TCP) traffic is considered "well-behaved" because it prevents the network becoming congested by means of closed-loop control of packet-loss and round-trip-time. The integration of both sorts of traffic is a complex problem, and depends on solutions such as admission control that have not yet been deployed on the Internet. Moreover, the problem of quality-of-service (QoS) and resource allocation is extremely relevant from the point of view of convergence of streaming media and data transmission on the Internet. In this paper an adaptive real-time protocol based on the least mean square (LMS) algorithm is proposed to estimate the application UDP bandwidth in order to reduce the quadratic error between the packet loss and a target. Moreover, the LMS algorithm is also applied to make sure that the reduction in the average bandwidth allocated to each TCP process will not be higher than a given percentage of the average bandwidth allocated before the beginning of the UDP application.

Recently, dynamic adaptive streaming over HTTP (DASH) has become an increasingly popular way to view video over the Internet. In particular compared to other video streaming services these DASH approaches deliver superior QoE to viewers. This is due to improved video segment selection. Generally, Transmission Control Protocol (TCP) CUBIC is the defacto transport layer protocol use by DASH. To improve the robustness of the transport layer to network congestion many other TCP variants were implemented such as Compound TCP and BBR. Nevertheless some were made to work specifically in LAN environments for example Agile-SD. However, recently another transport layer protocol User Datagram Protocol (UDP) has been used in Google’s QUIC implementation. To date no work has be found giving the performance of these transport layer protocols with DASH. In this paper we test the performance of Agile-SD, CUBIC, Compound TCP, BBR and QUIC using the BBA, MPC, Pensieve and Oboe DASH approaches. Experiments simulate congested bottleneck link conditions common at household routers where families view multiple videos at the same time. We observe Oboe and Agile-SD is the most promising combination with CUBIC and Pensieve next. However, even though QUIC was touted to have superior transport layer performance by Google it was the worst performing.

The object of research is Markov models of network nodes with UDP (User Datagram Protocol) and TCP (Transmission Control Protocol) traffic and their differences. The task solved is the lack of Markov models of network nodes describing the behavior of TCP traffic from the point of view of packet retransmissions and packet delivery guarantees. Markov models of network nodes describing traffic behavior with guaranteed packet delivery have been further advanced. Given the comparison of the models, the differences from the classic models serving TCP traffic were shown, for each packet flow, an additional dimensionally was added to the graph of states and transitions, which takes into account the retransmission of a lost packet. The comparison graph shows similar behavior of queue length and packet loss for both types of traffic. But the nature of the curves is different. With TCP traffic, packet loss can exceed 5 percent. In addition, lost packets must be retransmitted, which increases the load on the network node. More failures and packet queue lengths at a network node during peak load typically occur with TCP traffic compared to UDP traffic. At peak load, the difference in service failures can reach 20–30 percent. The main reason is that TCP uses flow control and rate-limiting mechanisms to avoid network congestion and ensure efficient data transfer between nodes. The Markov model of TCP traffic requires an additional dimensionally on the graph of states and transitions, which affects the behavior of queues and packet failures. The investigated problem was solved due to the universality and diversity of the mathematical apparatus of Markov mass service systems. The results could be used in network modeling software products for building and reengineering the topology of electronic communications networks at enterprises and organizations

An intermediate node in an inter-flow network coding scheme needs to know exactly which are the previous hop and next hop of a packet before coding. It is difficult to incorporate inter-flow network coding into opportunistic routing (OR) because the next hop of a packet in OR can't be determined in advance. Coding-aware opportunistic routing (CAR) is proposed in this paper to address this problem on fixed wireless mesh networks (WMNs). Meanwhile, it aims to maximize the number of native packets coded in each single transmission. It dynamically selects a route for a given flow according to the real-time coding opportunities. There are no control packets in CAR, which greatly reduces the overhead costs. CAR gives the coded packet that consists of a larger number of native packets with a smaller forwarding delay. The forwarder with the largest number of native packets coded together is ultimately selected to send data. Simulations demonstrate that CAR achieves significantly better throughput gains and derives a reasonable end-to-end delay in both cross topology and mesh topology under both transmission control protocol (TCP) and user datagram protocol (UDP) traffic, as explained below. CAR achieves more than 35 % throughput improvement under TCP traffic and more than 15 % throughput improvement under UDP traffic, compared to other state-of-art protocols in cross topology, respectively. CAR also provides a several-fold increase in throughput in a large scale network (mesh topology). In a word, CAR significantly improves network performance of a WMN.

One of the major problems in network games has been that of latency (lagging) that game technology researchers are still tackling. Latency largely affects user satisfaction and it is often caused by insufficient hardware capacity or the internet speed that the user is employing. Even though online games are becoming more complex and the number of participants in these games is increasing continuously, users cannot properly deal with the requirements to play these games, as system upgrades and subscription changes to a higher speed internet service are costly. Instead of passing such a burden on to the users, the game companies should instead invest in providing an improved communication algorithm. Thus, the reliable user datagram protocol (RUDP)-based communication system design has been considered in this research instead of transmission control protocol (TCP) or user datagram protocol (UDP)-based systems. This could be a viable alternative system model. Many researchers agree that RUDP is a better protocol for the transport layer, but it seems that the large-scale testbeds that could support such an idea and carry out direct tests are very scarce, meaning that actual experimental implementation is difficult to achieve. This suggests that game designers or researchers need to rely on a small-scale testbed to collect performance data. Moreover, generating an analytic model from a small-scale testbed may not be viable due to the large number of elements involved in the implementation process, including latency. This study introduces an RUDP-based communication model created and tested by using the OPNET simulation tool. They are novel energy and cost-effective photonics technologies for access/metro networks.

Learning platforms have become an integral part of the education system these days. Especially after the COVID era, education has become more allied with self-paced and remote learning and learning platforms have made it boom exponentially. Improper construction and implementation of such platforms can lead to huge risks for the users and the company. Data security is not taken much care of while building such platforms; instead, concentration is given to fancy front-end pages and attractive interfaces. This may not be good at all times. Data is one of the most powerful resources and can have a very big impact if misused. This paper proposes a networking-based approach to implementing such platform systems in a safe and organized way. Implementation using networking concepts gives a better hand in managing permissions, access rights, and security in all data-related transfers and communications. In terms of online gaming and real-time video communication, User Datagram Protocol (UDP) is often used because it is faster than Transmission Control Protocol (TCP) and is well-suited for real-time applications that cannot tolerate delays. UDP is a connectionless protocol, which means it doesn't retransmit lost data and therefore has lower overhead, making it a good choice for real-time applications like video conferencing and online gaming. Examples of such applications include Skype, Google Meet, Zoom, and Facetime. Based on these existing applications, this work introduces UDP in the field of Learning Platform Applications and builds a model on top of which real-time applications can be constructed. The proposed system makes use of UDP for all its requests, responses, and file transfers. The protocol itself is not very reliable, but the addition of provisions for acknowledgements in all requests and responses makes this system overcome transfer uncertainty. Implementation using networking concepts improves the speed, security, privacy, and customization abilities of the proposed system.

In this paper, the filtering issues are investigated for event-based Wireless Network Control Systems (WNCSs) with User Datagram Protocol (UDP). Because the optimal estimation of WCNSs with packet loss in both control and observation channels via UDP cannot be applied in practice due to the exponential increase of the computation, and in order to reduce the energy consumption of wireless communication nodes, an efficient recursive filter is developed for the event-based WNCSs with UDP. The state estimation and the filtering error covariance are derived from the interacting multiple model (IMM) methods, and the filter gain is determined by minimising an upper bound of the filtering error covariance. It is proved that the stability of the minimum upper bound of the filtering error covariance is only related to the actual arrival rate of the observation packet, and the minimum upper bound converges to the error covariance of the optimal estimation of the corresponding system with Transmission Control Protocol (TCP) if both the trigger threshold and the control input tend to zero. Numerical examples are given to verify the effectiveness and feasibility of the theoretical results.

On UDP continuity over vertical handovers

Due to the development and evolution in Cognitive Radio Networks (CRNs), the performance of Transmission Control Protocol (TCP) over CRN has become a crucial factor. TCP ensures the reliable end-to-end delivery of data, which is very important for the quality of service (QoS) requirements. TCP performance is severely affected by channel switching and channel availability in CRNs. The frequent occurrence of such events is due to CRNs nature which unnecessarily reduces the transmission rate which results in degrading the overall performance. In this paper, the authors investigate the limitations of using TCP NewReno (i.e., standard TCP) over CRNs, and further propose a cross-layer-design protocol based on an interaction between transport layer and lower layers to improve TCP performance over CRNs. The design is called Cross-Layer Design for TCP (TCP-CLD). It primarily aims at improving the TCP performance over CRNs via controlling and adapting a few selected TCP parameters. Specifically, the authors propose freezing the data transmission and re-transmission timeout (RTO) of TCP during the time of channel switching and channel unavailability. In addition, the authors update the size of congestion window (cwnd) of TCP based on the bandwidth and buffer size of relay nodes. Moreover, in the case of channel switching, the authors update the Round Trip Time (RTT) of TCP based on the new route delay. Simulation experiments have been conducted using ns-2 network simulator tool whereas results show that the proposed design achieves a significant performance improvement compared with the standard TCP NewReno protocol and the other known protocol, namely, TCP-CRAHN.

To deal with congestion control in internet protocol (IP) networks, different active queue management schemes have been successfully proposed. However, many of these schemes do not take into consideration the particular Voice over Internet Protocol (VoIP) requirements, thereby potentially penalizing this traffic. In this paper, we propose and evaluate a dropping packet selection algorithm for queue management schemes. This algorithm, called Drop-SwD , protects VoIP traffic without affecting other User Datagram Protocol (UDP) and Transmission Control Protocol (TCP) traffic. Additionally, to preserve the quality of the VoIP service, the proposed algorithm selects the lowest quality packets (with less impact on the subjective quality) to be dropped. The application of the algorithm in different active queue management schemes is evaluated using objective and subjective quality metrics and intelligibility metrics as well. As a result, we show that Drop-SwD improves the final quality of the voice packets’ flow, without degrading the performance of other traffic class, with respect to the packet dropping schemes of reference.