Beside that, spectrum efficiency also needs to be concerned to make sure that the future radio technology can be implemented. Spectrum efficiency is one of the important factors because of various digital radio technologies such as digital audio broadcasting (DAB), digital video broadcasting (DVB), Global System for Mobile Communications (GSM), General Packet Radio System (GPRS) and Universal Mobile Telecommunications Systems (UMTS) require large frequency spectrum. Since the radio spectrum is very limited, it needs to well planned and utilized so that other services can be introduced and implemented in the future. So, the Multimedia Broadcast and Multicast Service (MBMS) is one of the best alternatives in broadcasting technologies because of the efficient method in delivering multimedia content which at the same time, allows sharing of frequency resources between adjacent cells. This can be realized through SingleFrequencyNetwork (SFN) technique in indoor and outdoor area.
VI. C ONCLUSIONS
This paper extends the air interface of 3GPP 5G New Radio Release 15 to point-to-multipoint communications. The proposed solution, called 5G NR Mixed Mode, enables a flexible, dynamic and seamless switching between unicast and multicast or broadcast transmissions and the multiplexing of traffic under the same radio structures. Two 5G NR Mixed Mode operational deployments, Single-Cell Mixed Mode and Multiple-Cell Mixed Mode, have been envisaged for fulfilling the different 5G IMT-2020 usage scenarios, i.e. eMBB, URLLC and mMTC. The key principle design is to ensure the maximum compatibility with the current NR Rel-15 by reusing the original air interface and RAN upper layers as much as possible. The required modifications in the air interface include the introduction of a Group Radio Network Identifier, a multiple cell coordination for supporting SingleFrequency Networks, and narrower subcarrier spacings in order to allow larger inter-site distances. The proposed solution for the RAN upper layers involves the introduction of two new logical control channels (SC-MCCH and MC-MCCH) and two new logical traffic channels (SC-MTCH and MC-MTCH) as well as new RAN procedures that will enable the seamless switching between unicast and multicast control and data radio bearers. Furthermore, an enhanced Outer/Inner Loop Link Adaptation and 2 nd Layer FEC feedback schemes are proposed for the 5G
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Another major potential improvement for MBMS is the use of SingleFrequencyNetwork (SFN) operation of MBMS on the physical layer.
One way to increase MBMS coverage and to improve the radio link performance is the use of macro-diversity i.e., using several receiver and/or transmitter antennas for transferring the same signal. Being MBMS broadcasted simultaneously in different cells, network resources are not consumed when macro-diversity for MBMS is employed (this scenario does not occur when using a dedicated physical channel). Macro-diversity objective is to supply diversity in situations where the UE is far from the base stations, in order to compensate the transmission path loss of a terminal located at the edge of the cell or to increase the network capacity by reducing the amount of transmit power needed to reach a distant receiver.
mobile handsets. DVB-H was formally adopted as an European Telecommunications Standards Institute (ETSI) standard in November 2004[7, 23]. The two key features in DVB-H that were additional to DVB- T are: time-slicing and additional FEC, i.e., multi- protocol encapsulated FEC (MPE-FEC) – both at the link layer. The MPE-FEC provides better signal-to-noise ratio (SNR) and doppler performance in mobile channels for the MPE data. The time-slicing technique of DVB-H benefits in terms of power-saving of the mobile broadcast receivers, by transmission of data in bursts, thereby allowing the receivers to switch-off during the inactive period. At the physical layer there are four additions to the DVB-T specifications: 1) two additional bits in transmitter parameter signaling (TPS) indicating DVB- H service and MPE-FEC usage, 2) additional 4K OFDM mode included due to the trade-offs of mobility and SingleFrequencyNetwork (SFN) cell size and allowing receptions in medium size SFN at very high speeds, 3) in-depth interleaving bits over four or two OFDM symbols for 2K and 4K modes which improves tolerance for impulse noise, and 4) the 5 MHz channel bandwidth to be used in non-broadcast bands. DVB-H is backward compatible to DVB-T.
Figure 22. Delivery delays as a function of the distance for a group of 10 members
In Fig. 23 we depict the transmission delays under variable network load, and we consider a constant bit rate multicast traffic. We obtain these results when all the group members are located at a distance of 10 meters from the sender. We observe that the transmission delays increase significantly when the throughput exceeds the maximum capacity of the used protocol. This is because the buffering delays will be added. Besides, a packet is rejected when it exceeds the lifetime limit. Therefore, the highest delays are limited to this limit which is 60ms (as depicted in Table 8). Similar to the previous scenario, we observe that member DMS1 experiences lower delays than DMS10 when the throughput is up to 200pps. For the case of GCR-BACK, we notice that the delays are very limited when the packet rate is lower than 500pps. This is because a packet is immediately transmitted when it arrives to the MAC layer. Then the delays increase slightly for data rates from 500pps to 1500pps. This is because a packet may arrive while the protocol is in the feedback phase, i.e. the AP is exchanging BAR/BACK with the members. In this case the new packets wait the end of the exchange and the channel contention before being transmitted. When the throughput exceeds 1600pps, the delays of GCR-BACK increase significantly due to the buffering delays. This is because the highest supported throughput without queue overflow is limited to 1564pps, according to Fig. 17. However, we notice that the highest delays of GCR-BACK are much lower than the lifetime limit of 60ms. This is because these delays depend on the queue size; the packets are rejected if they arrive when the queue is full. But when a packet is in the queue, it should wait the transmission end of the older packets. In our case, the queue size is 20. In a saturated network, the delays of GCR-BACK are bounded by the maximum delay to send 20 packets, whenever this delay is lower than 60ms. We observe the same curve behavior for GCR-UR1, UR2 and UR3. However, GCR-UR3 reaches the saturation condition first. Besides, the maximum delays of UR3 are higher than those of UR1 in a saturated network. This is because the average service time for a packet under UR3 is more important than that required by UR1 and UR2.
The evaluators strongly agreed, as revealed by the computed mean =3.36 for the cost of research study mate- rials used are locally-available with standard quality and some of the materials and components are slightly used to be alternative source to attain reasonable costing. They also strongly agreed on the cost of the trainer unit as cheaper compared to the traditional FM broadcast station.
off the headlights while traveling at night. Nilsson and Larson  demonstrate how such an attack can be performed on the CAN protocol through simulation.
While gateways between internal and external networks might help improve security, it seems plausible that attackers will be able to circumvent or penetrate gateways and obtain the ability to send messages on an internal embedded network. Preventing such attacks requires strong authentication of nodes as an additional layer of protection. This presents a particular challenge in distributed embedded networks, because any authentication scheme must support multicast authentication subject to the constraints of: resource limited nodes, small packet sizes, potentially high packet loss rates, and tight real-time deadlines. We present an authentication method for distributed embedded networks which conforms to these common embedded system constraints. Our method allows the system designer to perform a tradeoff among per-message authentication cost, application level latency, and the probability of induced system failure. This is accomplished by appending truncated Message Authentication Codes (MACs) of only a few bits to each message. The time- triggered embedded applications we consider broadcast periodic updates of the values of system inputs and variables. This allows us to aggregate authentication from several messages before permitting an irrevocable alteration to the state of the system. Additionally, this approach allows us to reduce the probability of successful attacks on reactive control functions by making it difficult for attackers to forge enough messages in a short period of time to produce a system control failure.
In this the network node has 4 roles Sender, receiver, intermediate and jammer. Before selecting the jammer need to select sender and receiver. More than one intermediate nodes present in the network. When click the sender button, the sender form will appear in that need to choose the file which is going to transmit after chosen the file the path name in which the file located is shown. It contains a panel which maintains the intermediate nodes ip addresses. And then it shows sender ip that means from which node the file is transmitted and the ip address of the node going to receive the file and the port address. When clicking the send file button the status will be changed to sending from idle state. The frequency level is used when the intermediate chooses the UFH transmission. Before to send a file need to select the path in which the file will be saved after received. Then it contains ip address of sender, receiver and port address. This also contains filename and file size. After clicking start server listening the send file button has to click. The design of the intermediate node contains the name of the file going to transmit. And the mode of the intermediate node that is next to sender will be in receiving mode. Then it has three types of relaying method. First one is normal in which the data was successfully transmitted when there is no jamming. If there, the data will be blocked and communication will be failure. In UFH method the data was successfully transmitted if the jamming is present in the network. Because it forwards the data to multiple receivers. As well as the UDSSS also same but in this the file will be split and transmitted. In receiver side the split file will be merged.
Fig. 11. Video delay as a function of background traffic
that the measured QoS values do not stay within the required limits therefore causing connection requests being rejected, based on both bandwidth - and delay test. However, because the admission control 1 maintains the number of multicast receivers low enough all the time, the delay remains low throughout the simulation. This can be seen in figure 11, which depicts the delay experienced by the measured video customer. As the Qos requirements (jitter and packet loss) of video customers are a bit stricter than the ones of conference customers, it can be noted from figures 12 and 13 that video customers are being rejected more in relation to conference customers when background load increases.
MBMS broadcast mode may be used to deliver data items for individual users in a bandwidth efficient way. If similar data for every individual user is to be delivered separately, PDP context/SMS/MMS contexts are to be established for each individual user, which involves signaling overhead. No such overhead is necessary for MBMS broadcast mode. Individual user data items of similar nature and size are collated into a combined data stream. That stream is to be delivered using MBMS broadcast to multiple users. Individual user data is uniquely identified by a within the sequence number range. It is not bandwidth-efficient to include the sequence number explicitly along with each data. Hence there must be a status indication for every user in the form of a bitmap to indicate presence/absence of data. The status indication should be broadcast more often than the data stream itself. Each indicated in the status message, it should listen to a part of the combined data stream and be able to extract the corresponding data. A basic simulation model of MBMS broadcast should include simulation of a subset of MCCH and MSCH information. A socket programming setup using an iterative server with multiple client support is to be used to simulate the network and UE behavior. Data units are transmitted with a pre-defined clock rate from the server. Each unit includes a System Frame Number (SFN) count (from 0- 4095). Actual MCCH and MSCH messages are Abstract Syntax Notation (ASN.1) encoded. In absence of such encoder and decoder, only a subset to parameters have been chosen, and the structures themselves are transferred from server to UE client on designated System Frame Numbers (SFN). MTCH data is segmented into unacknowledged mode RLC PDUs. Those RLC PDUs are transmitted in the designated SFNs as per the scheduling information of MSCH data.
Abstract. Recently, it has been shown that applying MIMO technology, i.e. using multiple antennas at the transmit side and multiple antennas at the receive side, improves the performance of object detection and localization. In such scenarios, the spatial diversity specically helps overcome the fading of the cross section of the object, leading to reduced probability of missed detection. Such a phenomenon is, in fact, the dual of probability of bit error reduction in communication systems due to diversity gain. Despite the importance of such performance enhancement, this subject has not been suciently investigated in the PCL (Passive Coherent Location) schemes, where the transmitters (or illuminators of opportunity) used for localization are already present in the environment. Especially, in cases where the transmitters are working in a SFN (SingleFrequencyNetwork), such as the DVB-T (Digital Video Broadcasting-Terrestrial) signal, and all are transmitting the same signal, the situation becomes of higher importance. Obviously, the eect of the SFN environment invalidates the assumption of sending orthogonal waveforms traditionally used in localization schemes. In this paper, we design the Neyman-Pearson detector for a PCL scheme and show that we can achieve the desired diversity gain for such a design as well.
Until recently, the class of service attributed to multicast data packets, using UDP as the transport protocol, has been based on best effort without explicit flow control. As a result, data packets can be dropped due to congestion, or they can experience large variances in delay; both of which can cause poor performance in the application. Within an experimental environment, where the network is either over provisioned or has source thresholds (as evidenced in the Mbone), this type of framework can suffice for most one-to-many type of applications. But in going beyond today’s experimental environment, recent work has focused on designing and defining new classes of services [22, 23, 62, 87, 99]. Within the IETF, the term Integrated Services has been used to encompass two new classes of service for both unicast and multicast data: 1) Control Load, which focuses on providing a minimal level of end-to-end bandwidth, and 2) Guaranteed Service, which provides a minimal bound of end-to-end bandwidth and delay. The design of these services focus on data flows extending from host to destination(s) and does not address aggregation of services at a given edge; be it an administrative or topological boundary [45, 59, 111].
and mapping type. For each modulation order (constella- tion size), a single corresponding rotation has been chosen. In this section, we study the performance of rotated constellations for the previous different configurations used. More precisely, we will study the performance of rotated constellation for SFN-2x1 Alamouti MISO network with the configuration specified in Table 1 and TU6 channel with the configuration specified in Section 3. The results are shown in Table 4. This table gives the results of the SNR performance for different types of networks (SFN, SFN-2x1 Alamouti MISO) with and without rotated con- stellations over different numbers of antennas, and different types of constellations. Therefore, for instance in this table, the 256 QAM−2 means that the number of antennas is two and the constellation is 256 QAM. Note that the gain obtained by using a rotated constellation also depends on the receiver implementation. The worst case theoretically for the simplest implementation is supposed to obtain the same performance whether the rotated constellation can be used or not. These results show that for diversity transmis- sions, the values of SNR are almost stable with and without rotated constellations for all cases except for the TU6-MISO case where a deviation of 0.4 dB was obtained. For the pure SFN network case, the values of SNR depend on the type of modulation. For the 256 QAM modulations, the values are almost stable for rotated constellations in active and no-active modes. However, for others modulations (64 QAM, 16 QAM, QPSK) the deviation increases when bits per constellation decreases. For QPSK, the highest deviation was obtained.
Motivated by the expected lack of efficient multicast mech- anisms in manycore environments, we have evaluated the broadcast scalability of different WNoC schemes and com- pared it to that of aggressive NoC designs. The analysis considers full broadcast support in WNoCs through the integration of antennas on a per-core basis and the sharing of a single broadband channel among all cores. Besides en- abling the ordered delivery of broadcast traffic, this scheme provides a latency up to one order of magnitude lower than the best evaluated wireline counterpart. Beyond a few hundreds of cores and in spite of its much lower bisection bandwidth, WNoC attains a broadcast throughput commen- surate to that of conventional NoCs. For all this, we envisage a hybrid network architecture where a WNoC will serve broadcast traffic and a conventional NoC will transport the rest of communication flows. With such scheme, the latency is reduced dramatically for high levels of broadcast, whereas the throughput is significantly increased for low levels of broadcast. The improvement becomes more patent as the system size increases, ensuring the suitability of such hybrid approach in the manycore scenario. To achieve such goal, though, we stress the need of a channel capacity commensurate to the rate at which cores can inject data, as well as of a flexible and reasonably efficient MAC protocol. The latter requirement can be either met with common MAC protocols or amply exceeded by virtue of protocols that take advantage of the unique optimization advantages of the multiprocessor scenario.
In this paper, we propose a new hybrid multicast scheme, PAM, which as opposed to native IP multicast, does not require
all routers to be IP multicast-enabled, and as opposed to ALM,
does not exclude network support. Instead, PAM relies on partial network support, selects a small subset of routers as PAM-enabled multicast routers that are strategically located to serve group communication, and adapts its selection based on group dynamics. As a result, PAM, as opposed to existing hybrid multicast schemes, is suitable for both sparse and dense communication groups. PAM also reduces the network overhead inherent in native IP multicast, and does not suffer the delay stretch and the high stress inherent in application- level multicast. Experimental results on both synthetic and realistic Internet topologies, for both sparse and dense groups, reveal that PAM can achieve efficient group communication with no delay stretch, an average stress of merely 1.25, while using less than 15% of the multicast routers that are needed in native IP multicast.
Wireless technologies have revolutionized the world of communications. It started with the use of radio receivers or transmitters for use in wireless telegraphy .The term wireless are used to describe technologies such as the cellular networks and wireless broadband Internet. Although the wireless medium has limited spectrum along with a few other constraints as compared to the guided media, it provides the only means of mobile communication. Wireless networking is used for random and rapid deployment of a large number of nodes, which is a technology with a wide range of applications such as tactical communications, disaster relief operations, health care and temporary networking in areas that are not densely populated . A mobile ad-hoc network (MANET) consists of mobile hosts equipped with wireless communication devices. The transmission of a mobile host is received by all hosts within its transmission range due to the broadcast nature of wireless communication and Omni-directional antennae. If two wireless hosts are not within the transmission range in ad hoc networks , other mobile hosts located between them can forward their messages, which effectively build connected networks among the mobile hosts in the deployed area. The use of wireless ad hoc networks also introduces additional security challenges that have to be dealt with. Compromised node and denial of service are two key attacks in wireless sensor networks .In this proposal, data delivery mechanisms that can with high probability circumvent black holes formed by these attacks. The routing algorithms namely Non-Repetitive Random Propagation (NRRP), Directed Random Propagation (DRP) and a new Optimized algorithm is proposed which would reduce the route discovery time and no of hops from source to destination. A Zone Leader multicasting Algorithm is also implemented for multiple zones. The challenge faced nowadays is to design a robust multicast routing protocol for a scalable wireless network. The increased quantity of data transmission and reception in wireless networks has adversely increased the need for bandwidth on demand and quality of service. The further increase in data traffic, leads to loss of information, accuracy and reliability  . To overcome this drawback, we have proposed a scalable and efficient multicast optimized algorithm.
In a typical data multicast transmission from AP to a lot of nodes, the nodes are divided into several clusters. As depicted in Figure 1, our system model is the scenario where the AP broadcasts the packets to a single cluster, which consists of a cluster head (CH) and K common nodes (CNs). The cluster head takes responsibility to deliver the packet to common nodes in the cluster. Namely, common node can not communicate with the other common nodes and communication links only exist between the CH and CNs. The AP can be considered as an unmanned aerial vehicle (UAV) or stratospheric tele- communication platform, which conveys the information to the nodes on the ground. Due to high signal attenua- tion, communications from the AP to the nodes suffer from high loss rates. However, the communications among the nodes on the ground always experience good channel quality. So the nodes on the ground can cooper- ate together to recover lost packet locally.
There are two protocols for host naming on a local network without a central DNS server: Multicast DNS (mDNS) and Link-local Multicast Name Resoultion (LLMNR) . LLMNR is currently implemented only in Windows Vista and Windows CE, but mDNS is implemented on Windows, Linux and Mac OS, and has been successfully ported to other POSIX platforms and Java-based platforms  as well. Multicast DNS enables the translation between a local host name and IP address without a central DNS server. This local host name selected by each device is meaningful only on the local network. Multicast DNS service discovery  allows users to announce their services and discover peer services. The service discovery protocol uses three record types: PTR, SRV and TXT record. The PTR record is used to discover service instances on the local network. The SRV record provides port number and IP addresses of the services. The TXT record provides additional information about services.
For example, consider the following scenario in streaming- media delivery via multicast. During an end-hosts multi- cast session, a large fraction of clients (that were acting as routers) might unsubscribe together in the middle of the stream, partitioning the overlay network. The time taken to repair the partitions will result in loss of packets transmitted during the transience period. Such an event, while probable in end-host multicast, is in contrast to network-level mul- ticast where clients only occur as leaves in the multicast tree built on routers. Router failures are not as frequent, and the chances of simultaneous failures of a large num- ber of routers, occurring frequently over the stream dura- tion, is extremely small. Hence, while packet losses due to router failures is not a probable eventuality in IP-Multicast, it must be accounted for in an evaluation of an end-host mul- ticast proposal. Unfortunately, conventional evaluations of such proposals have glossed over such implications on end- application performance.
The concept of network coding was introduced in 2000 . Different from the store-and-forward scheme, network coding allows intermediate nodes to recombine packets received from different incoming links. This paradigm is featured with a number of significant advantages, such as balanced network payload, robust security, strong resistance to network failures, energy-efficient transmission, and so on. In addition, when applied to multicast, network coding can always achieve the theoretical maximum throughput. This makes network coding based multicast (NCM) an ideal technology to support one-to-many broadband data transmission . Therefore, NCM has received a significant amount of research attention from areas, such as information theory and computer science, for more than one decade .