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Departing from the well-known problem of the excessive overhead and latency of connection oriented protocols, this paper describes a new almost reliable connectionless protocol that uses User Datagram Protocol (UDP) segment format and is UDP compatible. The problem is presented and described, the motivation, the possible areas of interest and the c...
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... Whatsapp or Facebook) could automatically assess or infer the quality of experience of any given call. For example, after closing a Facebook Messenger video call, the application shows a feedback dialog window where the user is expected to rate how well the communication was experienced, and what the user felt went wrong (please see Figure 1 and Figure 2). ...
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... Whatsapp or Face- book) could automatically assess or infer the quality of expe- rience of any given call. For example, after closing a Face- book Messenger video call, the application shows a feedback dialog window where the user is expected to rate how well the communication was experienced, and what the user felt went wrong (please see Figure 1 and Figure 2). For all of the above examples, but also to other examples where some type of data imputation [4] could be done to e.g. ...
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... UDP is a 'connectionless' protocol that operates at the transport layer [16]. The length, beginning of the port location, objective port location, and the checksum fields are all 8 bytes in a User Datagram Protocol header. ...
Link failure refers to the failure between two connections/nodes in a perfectly working simulation scenario at a particular instance. Transport layer routing protocols form an important basis of setting up a simulation, with Transmission Control Protocol and User Datagram Protocol being the primary of them. The research makes use of Network Simulator v2.35 to conduct different simulation experiments for link failure and provide validation results. In this paper, both protocols, TCP and UDP are compared based on the throughput of packets delivered from one node to the other constrained to the condition that for a certain interval of time the link fails and the simulation time remains the same for either of the protocols. Overall, this analysis is based on determining the performance of both protocols with a fixed packet size and bandwidth. This analysis, performed with the help of NS2 and XGraph, shows that the transport layer protocol, UDP acts better than TCP in terms of throughput. This opens the questions to other fellow researchers of how different metrics act in both the cases when a link failure occurs. In UDP, the throughput drops less as compared to the TCP at the time of the link failure regardless of if simulation was executed for different time periods i.e., 70,100,300,900 and 1000 seconds. The link failure interval is also varied from 10,15,20,40,350 and 440 seconds to generalize and validate the performance of the network during the interval.
... The consequences of the usually termed "high overhead", "slow" TCP versus the "low overhead" and "light" UDP protocols are a well-studied subject and have given many contributions to network science, including e.g. many different "flavours" of TCP protocols, such as Reno, Vegas, among others [61]- [64]. But however well researched a subject is, there is always room for innovation, and this was what the concept of the Keyed User Datagram ...
... Protocol (KUDP) [64] proposes. ...
... It is not the goal of this section to fully describe the concept and operation of KUDP, as this is best described in [64]. But a brief introduction has to be done to further extend the concept of Keyed IPv6 presented in section 5.3. ...
The network layer is central to the networking scientific area. It is around the network layer that all the data communications develop, and one of its main tasks is to allow the identification of each single interface/machine between the potentially many interfaces in a network. This seminar addresses some of the issues that are usually presented to young Computer Science Engineering students in the course of several classes, but also presents some topics that are not address in networking courses. It is mostly focused on using Internet Protocol addresses in Local Area Networks, also considering issues that belong to the Wide Area Networks, such as data aggregation. This document summarizes the content of a seminar, therefore it comprehends both teaching and researching subject. The seminar starts with a history of the evolution of the communication protocols from the early days of networks up until IPv6. It describes a new approach to define the addresses of network interfaces using Variable Length Subnet Masks, as usually this is a not an easy task for Computer Science Engineering undergraduate students. This summary also describes some of the limitations of the data communication in todays' networks, proposing some solutions, where possible, including a novel mean of connectionless data transmission by using IPv6 addresses, by extension of previously published research. The way the seminar is organized provides a history to the past of the Internet Protocol, a view of some of its well-known current limitations, and a glimpse into a possible future regarding an improved connectionless layer 3 data transfer protocol.
... Authors in [3] have proposed a new Keyed UDP protocol, in which the use of different port numbers, in a sequence, allow for the destination machine to perceive, to some extent, how successful the transmission was, including, what were the packets that were lost, and also to reorder packets that were received out-of-sequence. While the description of the base concepts for the Keyed UDP are described this recent paper [3], its authors did not provide simulations that proved the advantages of the new protocol. ...
... Authors in [3] have proposed a new Keyed UDP protocol, in which the use of different port numbers, in a sequence, allow for the destination machine to perceive, to some extent, how successful the transmission was, including, what were the packets that were lost, and also to reorder packets that were received out-of-sequence. While the description of the base concepts for the Keyed UDP are described this recent paper [3], its authors did not provide simulations that proved the advantages of the new protocol. Please note that Keyed UDP does not imply a change in the UDP format [2] and [3] foresees compatibility between applications using UDP and Keyed UDP. ...
... While the description of the base concepts for the Keyed UDP are described this recent paper [3], its authors did not provide simulations that proved the advantages of the new protocol. Please note that Keyed UDP does not imply a change in the UDP format [2] and [3] foresees compatibility between applications using UDP and Keyed UDP. This paper presents, in a very condensed manner, the explanation of the basics of the Keyed UDP and the first results obtained by simulation regarding different configuration parameters for the Keyed UDP, including varying loss ratios and ratios for packets received out-ofsequence, and different key lengths for the protocol. ...
The User Datagram Protocol (UDP) and other similar protocols send the application data from the source machine to the destination machine inside segments, without foreseeing nor allowing for any type of control on the transmission or success metrics. These protocols are very convenient for e.g. real time data transmission. But when the reliability of the transmitted data is critical, other protocols termed as connection-oriented, allow for full control of the data transmission process, assuring that the received data is an exact copy of the transmitted data, e.g. the case of the Transmission Control Protocol (TCP). To sustain the increased functionality and features of the connection-oriented protocol, a set of mechanisms is implemented based on some specific fields of the segment header. These mechanisms result in a significant overhead in terms of the increased number of transmitted packets. This may further translate into significant delays, because of the additional number of switching and routing tasks, and eventually, because of more complex communications procedures, such as e.g. transmission window resizing, and of course, acknowledgement and sequence numbers updating. The two extremes of these communication modalities, one that has no control at all, and the other one that allows for full control, have resulted in the creation of an intermediate protocol that allows for a limited degree of knowledge on how successful a transmission was, and even for an eventual reordering of the segments that arrive out of sequence. This paper presents simulation results that confirm the efficiency of the new almost-reliable UDP protocol, termed Keyed User Datagram Protocol (or KUDP) for transmission of data that includes the ability to identify which packets were lost and to reorder packets that were received out-of-sequence, and points future tasks to be pursued in this research.
... We term situations where MNBC i < 20 L as very tiny buffers. Since a typical subnetwork will consist of as a minimum, two nodes (as a node cannot be standing on its own in the Internet-it must, in the context of guided media-linked Internet, be connected to at least one other node), and a maximum, which depends on the structure (topology) of the subnetwork: using the extended Ethernet frame length (12 240 bits), shown in Figure 10 (Ethernet-based networks now operate on switching principles), 21 we present in Table 2 specified buffer capacities for star point node, for subnetworks consisting of two to five nodes. Subnetworks with more than T A B L E 2 MBCs computed for star point nodes for subnets with two to five nodes, for some σ = xL, using Ethernet frames ...
Switched networks' (including the Internet) nodal devices require buffers to hold packets during periods of congestion and when traffic streams on more than one input port need to access the same output port. Appropriately sizing the buffers has been a contentious issue among researchers over the last couple of years. Three main buffer sizing formulas have been reported in literature, which are: bandwidth‐delay product or BDP, small buffer, and the tiny buffer formulas. But researchers are largely agreed that the BDP formula results in unrealistically large and wasteful buffers, while a number of researchers have held with good reasons, that the small and tiny buffer formulas are theoretical formulations that cannot be generally and practically applied to real networks. Some researchers have even asserted that deriving a single, universal formula for sizing buffers may not be possible. The main purpose of this paper is to explain an approach, which we term “network topology point –of –view,” that we developed to derive a novel, empirical, generic, practical, easy‐to‐apply, closed‐form formula that can be used to obtain the buffer capacity that is optimal—in terms of the minimum capacity possible for lossless operation, for any switched network node, including nodes (switches and routers) in the Internet. The results obtained from utilizing the formula to specify typical nodal buffers' capacities are largely in agreement with the values of the tiny buffer formula in literature; but this paper's formula additionally specifies what we term “very tiny buffers.” The simple approach adopted for presenting this paper makes it a handy tutorial material that explains this “network topology point‐of‐view” concept.
In the present work a modeling of an IP - based corporate network with different traffic loads - Internet browsing, video conferencing, working with databases and VoIP calls has been done. The purpose of the modeling is to check the convergence time of the modeled network for different traffic loads using different routing protocols. Modeling is done for the following protocols: RIP, EIGRP, OSPF, OSPFv3 and RIPng. Additionally, modeling was done when part of the links between the branches in the network fails and after some time recovers. This is done to determine what would be the convergence time when the network topology changes during operation.
