On UDP continuity over vertical handovers

Mobile data offloading using unlicensed WiFi spectrum represents one of the innovative approaches that can be adopted to cope with the huge amount of mobile traffic demand. However, the main challenge in such approach lies in the ability of the system to discover the WiFi coverage for users who may need to offload their data. For this, the access network discovery and selection function (ANDSF) has been introduced by 3GPP to provide the mobile user equipment (UE) with the discovery information and the list of available WiFi hotspots. However, the current ANDSF discovery mechanism depends on the geographical location of the UE, which need to be accurately determined and continuously transmitted to the ANDSF server. In this paper, an enhanced ANDSF WiFi discovery technique is proposed and validated. The proposed technique is designed to use the reference signal received power (RSRP) information measured at UEs from their surrounding cellular base stations to build fingerprints for various WiFi hotspot areas. Based on the constructed fingerprints, the UE's WiFi coverage state can be identified using the proposed decision tree (DT) machine learning approach. The conducted tests and generated results validate the operation and performance of the proposed machine learning-based discovery technique with accuracy reaches up to 95%.

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 propose an improved IP-based handover scheme, named the Heterogeneous Network Handover by using 3GPP ANDSF (HNH3A for short) for S-PMIPv6 where S-PMIPv6 is a MIPv6 family protocol developed in one of our previous studies, and ANDSF (standing for Access Network Discovery and Selection Function) is an entity within an evolved packet core (EPC) of the system architecture evolution (SAE) for 3GPP compliant mobile networks. The purpose of using the ANDSF is to assist user equipment (UE) to discover non-3GPP access networks, such as WiFi or WIMAX. The HNH3A can help the handover among mobile WiMAX (i.e., Worldwide Interoperability for Microwave Access), 3GPP (i.e., 3rd Generation Partnership Project) family (including 3GPP systems, 3G LTE and 4G LTE-A) and WiFi based on existing handover techniques among the three heterogeneous networks. The analytical results demonstrate that the HNH3A can effectively mitigate handover delays and handover signaling costs.

This chapter considers the challenges faced by network operators and service providers accommodating the increasing traffic demands in cellular networks in the most efficient yet inexpensive way. It proposes that these challenges are addressed by offloading part of the traffic to femto cell access points (FAPs) and WiFi access points (WiFiAPs). Whereas 4G micro and pico cell base stations are assumed to be managed by the network operator in terms of setup and maintenance, FAPs and WiFiAPs are normally bought and operated by the end-user. The main difference between these two solutions is that the FAPs operate on the frequency bands assigned to the network operator by national regulators, while WiFiAPs work on unlicensed spectrum. This chapter analyzes the pros and cons of such approaches, and the tradeoffs related to the different mechanisms employed in cellular and WiFi networks for interference management. Moreover, various methods for traffic offloading by the mobile network operator are discussed in detail, including local IP access (LIPA), selective IP traffic offloading (SIPTO), IP flow mobility (IFOM), access network discovery and selection function (ANDSF), and Hotspot 2.0, etc. In the experimental part of this chapter, means for traffic management from the network operator's perspective are discussed, taking into account costs and energy savings. Furthermore, a novel resource usage coordination concept in conjunction with the WiFi offloading concept is presented. Introduction The traffic served by cellular networks grows significantly on a yearly basis, making the efficient, fair, and inexpensive management of users’ traffic demands a real challenge. Various forecasts provided both by legal bodies and commercial companies show that this trend will not change in the future. One can, for example, consider the expectations provided in Table 6.1, created based on the data delivered in the Cisco Visual Networking Index [1, 2] in 2012 and 2013. It is shown that the monthly average traffic generated by mobile devices is foreseen to grow exponentially. Based on [3], it can be stated that in 2017 IP traffic originating from mobile devices will comprise more than 13% of fixed IP traffic, while in 2013 this comprised only around 4% of fixed IP traffic.

Network Offloading schemes play a vital role when there is a massive growth in the capacity of Next Generation Networks. The aggregation of LTE (Long Term Evolution) and WLAN (Wireless Local Area Network) is meant to provide service with seamless connectivity for the User Equipment (UE). There are two major issues that have been identified in existing RSSI (Received Signal Strength Indicator) based offloading scheme such as: i) it fails to provide better performance ii) it requires frequent reconnection for establishing dual connectivity. UE (User Equipment) based offloading approach requires a lot of signaling between UE and Radio access network and thus lacks in resource utilization and energy efficiency. Network based offloading approach needs a network specific entity. In the proposed LWA-SA (LTE WLAN Aggregation-SDN Assisted) which is an SDN (Software Defined Networking) based offloading scheme. In this, a controller keeps track of ongoing sessions after offloading. Here, an algorithm is used based on Access Network Discovery and Selection Function (ANDSF) offloading scheme for the purpose of selecting an efficient Wi-Fi access point which increases the throughput and reduces the delay and also provides seamless connectivity to the UE.

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.

Context-aware, user-driven, network-controlled RAT selection for 5G networks

To improve the interworking between LTE and WiFi wireless technologies, an efficient handover mechanism is proposed. Dynamic and critical measures like mobile-assisted parameters are inserted to the Access Network Discovery and Selection Function (ANDSF) entity defined by 3GPP standard. Moreover, Context Aware (CA) decision algorithm is integrated within the ANDSF server so that the inserted measurements are expeditiously invested to improve the handover decision policy. The performance enhancement of the proposed model compared to the traditional one is shown by numerical simulations.

<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>

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.

Software Defined Networks (SDN) is a networking paradigm that separates the control plane from the data plan of a network, and controls data flow with a SDN controller. This networking model would be useful for ISP (Internet Service Provider) networks in order to control their own networks via the controller. Unfortunately, it is difficult to fully implement SDN in ISP networking due to financial and operational constraints. In this article, we will identify the various migration challenges for hybrid SDN in ISP networks, and investigate on the network planning tools currently available in the market that with regards to facilitate the migration process. Lastly, we will discuss on the future direction of research and development in the field of hybrid SDN migration.

In this paper, we present preliminary results of our ongoing networked virtual reality project by discussing the implementation and performance of a networked haptic game. In this game, online players feel like handling a real basketball since they are able to feel the sense of touch by using haptic interfaces. One of challenging issues in this implementation is haptic data transmission over the Internet to allow online multi-user play. Since haptic information is extremely sensitive to delay and jitter, the provision of timely transmission is critical. We carry out experiments to compare the performance of the haptic data transmission when using UDP (User Datagram Protocol) and TCP (Transmission Control Protocol), and observe that neither UDP nor TCP is able to satisfy QoS (Quality of Service) requirements of the haptic transmission. As a further study, we plan to investigate some alternative haptic data transmission schemes.

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