Scalability and performance are major challenges in developing high-performance multicast packet switching technologies in the 5G/6G core networks. We propose a novel scheme called 2-D scalar-matrix and vectors routing and forwarding (2D-SVRF) to address these two issues. Our scheme improves upon commonly used algorithms, Bloom filter (BF) and scalar-pair vectors routing and forwarding (SVRF), which have limitations in space and time efficiency, especially in carrier-grade packet forwarding engines (PFEs) with high-port-density and large membership-capacity. 2D-SVRF transforms a scalar-vector into a scalar-matrix by dividing an n-element group into N-rows (sub-blocks) and dividing an output-port bitmap with$\rho $-elements into M-columns within each sub-block. This enables the reuse of smaller and identical prime keys among different sub-blocks and subscalars. By leveraging this approach, multicast forwarding can be partitioned to exploit parallelism, resulting in reduced memory usage and lower computational complexity. Simulation evaluation confirms that 2D-SVRF outperforms competing algorithms in terms of scalability and efficiency, particularly when optimized with the (M, N) parameters. The 2D-SVRF approach is expected to make the implementation of SVRF feasible in carrier-grade multicast-enabled switches and routers.

As a key infrastructure of Industry 4.0, Industrial Internet-of-Things (IIoT) promises the opportunity to build powerful industrial environments by leveraging the growing ubiquity of wired and wireless communication technologies. Designing a data delivery scheme in the future IIoT networks is undoubtedly a challenging task, as it should satisfy several conflicting requirements: massive-scale, data-intensive, and mission-critical, the requirements of which have motivated the desired need for feasible IIoT network architecture. In particular, the traffic characteristics of certain IIoT applications feature small-data patterns especially in typical automation control scenarios such as robot control on the downlink. In order to reduce the huge overhead associated with each individual unicast transmission of the small-data message, we propose a novel “multipoint-to-multipoint” and/or “point-to-multipoint with different contents” communication paradigm-Uni-Multi-Unicast (UMUcast), which is based-on traditional 4G technologies such as evolved Multimedia Broadcast Multicast Service (eMBMS) and Group Communication System Enablers (GCSE), and novel 5G technologies such as Multi-Access Edge Computing (MEC). For the downlink, the UMUcast transmitter can jointly encode multiple-sources' small-data messages into a single chunk at MEC equipment in conjoint with a gNB, where chunk can be one-off transmission to multiple receivers simultaneously through an eMBMS frame, whereas the chunk is decoded into multiple small-data by each individual IIoT devices separately and respectively. The simulation results show that UMUcast has remarkable improvements over conventional point-to-point unicasting in handling multi-source, multi-destination, and massive small-data delivery with characteristics of low-overhead, high-throughput, and ultra-low-latency for future 5G-based IIoT networks.

Due to the advances of VoIP services such as Skype and Line, people are able to make conversation over the Internet with very low cost. However, VoIP is a real-time application and large end-to-end delay may cause poor communication quality. Most existing mechanisms to improve VoIP QoS are based on the traffic classes, but various voice traffic flows may experience different delays due to different end-to-end distances, and hence various number of intermediate hops which cause variation of delay. In this paper, we propose a novel mechanism that assigns different VoIP flows into different queues based on the path selection by SDN. Usually the biggest variable delay in IP voice communication systems is the queuing delay introduced by switches or routers along the path, and the queuing delay is influenced by the link utilization. Thus, we use hop-count and link utilization to determine which queue the incoming VoIP flows should go through. To ease the task of calculating hop-count of flows and link utilization in the intermediate nodes, we use software-defined network (SDN) to perform the task of routing path selection. The simulation results show that our proposed mechanism can improve QoS of VoIP significantly.

Nowadays many IPTV services are deployed based on P2P structure for the sake of saving the cost of server equipment and bandwidth requirement. Since peers may join and leave the P2P system arbitrarily, it may cause the service quality unstable. In this paper, we compare IP addresses to find partner peers in proximity, and use specific packet pairs to probe partner peers that have larger bandwidth, so as to reduce the switching delay thru pushing the video chunks proactively. We also use two buffers that store data of both current and newly selected channel to improve the stability of media playback. We use OMNET++ simulator to evaluate our proposed schemes, the numerical results show that our schemes significantly reduce the channel switching delay and improve the smoothness of video playback.

The continuous growth in traffic volumes on modern Data Center Networks (DCNs) can be classified into either delay-or throughput-sensitive applications. In order to providing quality delivery to such applications, it is important that multipath routing and/or forwarding can be supported in the DCNs. However, some inherent problems of multipath forwarding have not been totally solved by existing solutions. Aiming to balance all these requirements for the multipath DCNs, we propose a novel solution - Code-Oriented eXplicit MultiPath (COXMP) scheme which evenly distributes traffic along multiple paths, resulting in better network throughput, delay, jitter, and resilience. The simulation results show that COXMP outperforms existing solutions and achieves a reasonable tradeoff between scalability and overhead.

Live streaming (a.k.a. Internet Protocol Television (IPTV)) services have been popularly deployed on the Internet. Traditionally, IPTV service provider adopted client-server architecture. However, when the client population grows up to a certain amount, it usually faces the scalability and cost-effective problems. Fortunately, the development of Peer to Peer (P2P) content sharing application has proved that the P2P paradigm is an efficient solution to deliver many kinds of content over the Internet. P2P IPTV services may face the problem that there are more and more Network Address Translator (NAT) users, and peers behind NAT (NATed peer) cannot directly deliver its streaming data to other NATed peers. This situation easily leads to the so-called free rider problem. In this paper, we propose a grouping and scheduling scheme to increase the share rate of NATed peers. By restricting transmission connectivity of those peers with public IP address and properly arrange data chunks delivery, we could allow NATed peer to contribute more data on the overlay network. Simulation results show that proposed scheme can improve the performance in terms of the share rate and video playback quality compared to original NAT traversal mechanism. Besides, the deployment complexity of both P2P IPTV operators and peers, are expected reduction.

Explicit multicasting protocols such as Differential Destination Multicast (DDM) in MANETs have been retaining advantages of traditional approach while eliminating their shortcomings especially for small-group applications. However, they still have some drawbacks especially in the group size limitation. In this work we discussed a new unified unicast and multicast routing scheme called Arithmetic Labeled EXplicit (ALEX) Multicast Routing based on arithmetic encoding of on-tree node keys and recursive unicast for MANETs. Our scheme constructs the multicast tree by decoding the bitmap of its output ports for next hops in the forwarding path using a common Multicast Channel IDentifier (MCID) and a node-specific key, so that multicast packets can be self-routed to multiple receivers without requiring header modification, and intermediate nodes on the multicast mesh will be stateless. In addition to scaling up the multicast group size, our scheme can reduce the processing complexity, control overhead and delivery latency comparing to the existing works. The simulation results indicated that our proposed scheme achieves a remarkable improvement in scalability while simplifying the deployment and management for a large number of medium-size multicast groups in MANETs.

In this paper, we apply IEEE 802.1Qbu Preemptive Priority Frame Forwarding mechanism to the IEEE 802.3z Gigabit Ethernet networks, and compare its performance with non-frame-preemption queuing schemes for various traffic patterns. A feasible implementation of the proposed solutions is addressed. The FPF effectively reduces the forwarding delay as well as jitter, so as to enhance response time of real-time services in GbE networks. Through simulations, we show that the FPF achieves superior and precise QoS control compared to existing non-preemptive-based priority scheduling schemes.

A novel label switching architecture for all-optical label switching (AOLS) networks is firstly proposed and experimentally demonstrated. Proposed scheme adopts successful factors of existing architecture such as GMPLS, and to construct a high performance, scalable and manageable IPbased AOLS network. Similar to GMPLS, a particular label based on source routing information is attached to IP packets/burst for fast interconnection purpose rather than an option as source-routing in conventional IP framework, thus the scheme requires neither table lookup nor header replacing in the intermediate node, labeled packets will be quickly forwarded through self-routing, and the only operation in switching nodes is to strip off its part of the label which represents the output port, which eliminates the O/E/O conversion as much as possible. The benefits include simple switching process with low packet delay, higher security, and better scalability on next generation optical Internet.

IEEE 802.16j mobile relay station (MRS) is used to extend the coverage area of WiMAX. The MRS provides a fixed access link to mobile terminals riding on a vehicle. When a mobile station (MS) moving together with MRS changes its associated base station (BS), it performs network layer handover while MRS performs link layer handover. It usually causes a long service disruption time (SDT) that seriously affects the real-time mobile services such as VoIP. In this article, we proposed a pre-authentication scheme that reduces the disruption time of link layer handover by performing the authentication procedures before the MS roams to another BS. Through both analysis and simulation, we show that with Fast Mobile IPv6, our proposed scheme is able to reduce the SDT significantly in both predictive mode and reactive mode. Therefore it is able to improve the quality of service (QoS) for real-time mobile services.

Client-based Mobile IPv6 (MIPv6) is the most well known mobility management protocol, and the fast emerging proxy-based Mobile IPv6 (PMIPv6) offers an alternative. Some inherent problems such as route optimization in these protocols have not been really solved. Although various proposals tried to tackle the route optimization problem, none of them has achieved a real success. Furthermore, most of them are not comprehensive solutions for coexisting MIPv6/PMIPv6 mobile environment. In this paper, we proposed a novel unified route optimization scheme - Traffic Driven Pseudo Binding Update (TDPBU) to accommodate various mobility management issues. We evaluated our proposed scheme through simulations, and numerical results show that TDPBU can significantly improve the overall performance of mobility management protocols. Index Terms—Route Optimization, Mobile IPv6 (MIPv6), Proxy Mobile IPv6 (PMIPv6), Traffic-Driven Pseudo Binding Update (TDPBU), Network Mobility (NEMO).

Due to the increasing volume of multimedia traffic on the Internet nowadays, the traditional client-server architecture may face scalability and deployment cost problems when the number of subscribers grows large. To tackle these problems, more and more media streaming services, such as IPTV are deployed based on Peer-to-Peer (P2P) architecture. According to the current Internet infrastructure, most of subscribers are behind Network Address Translator (NAT) and using asymmetric access link, so they can not directly share their data with other NATed peers. This situation can easily lead to insufficient upload bandwidth and result in video stuck. Further, the improper selection of peers to download video data may also slow down the video delivery. Both above situations degrade the quality of services of P2PTV system. In this article, we propose approaches to deal with peer grouping and source peer selection, the former is to restrict connection to certain peer groups for improving the share rate of NATed peers, and the latter is to probe partner resource status for optimizing data scheduling. Simulation results show that our proposed approach is able to increase the share rate of NATed peers as well as to reduce video downloading time significantly.

Recently, the Orthogonal Frequency Division Multiplexing Access (OFDM/OFDMA) based wireless transmission technology has been widely accepted by consumers. Frame Check Sequence (FCS) scheme was employed to enhance the reliability of OFDM/OFDMA systems. Since the padding overhead cannot be effectively avoided in OFDM/OFDMA when MAC frames are encapsulated. We propose a novel CRC based error correction scheme by utilizing the frame padding to carry extra segmented FCS information as much as possible; compared with legacy FCS scheme, our approach obtains a significant improvement in error detection and correction upon the first retransmission (2ndtransmission). In addition, we demonstrate a significant performance improvement based on the mathematic analysis.

Internet Protocol Television (IPTV) is an emerging and killer application for WiMAX networks, which takes advantage of multicasting mechanisms to efficiently deliver the video data to groups of subscribers. In this paper, we propose a Reception-rate-tracking Opportunistic Multicast Scheduling (ROMS) to enhance the performance of scalable IPTV layered multicasting over WiMAX networks. The proposed ROMS jointly considers the subscriber scheduling and resource allocation to achieve resource efficient multicasting. We validate our solution through theoretical analyses and empirical simulations. Our study indicates that the proposed solution outperforms the Slot-throughput-tracking Opportunistic Multicast Scheduling (SOMS) in terms of resource efficiency for 19.6%-33% in different subscriber group sizes.

This paper gives a theoretical approach to modeling and simulating the Grade of Service (GoS) of Voice over Peer-to-Peer Session Initiation Protocol system which is created based on various teletraffic parameters. VoP2P features a particular service characteristic due to its different GoS models originated from circuit switched telephone networks. Especially, most voice session in VoP2P is indirectly connected because peer-nodes usually locate behind NATs. This situation forces peer-node to select one or more relay-node for establishing sessions. However, the relay-node is a constrained resource because every relay-node can only support very limited relay services in VoP2P network. There are two strategies, the S2RS (Staged Single-Relay Strategy) and MRS (Multi-Relay Strategy), to solve the relay-node issue. We adopt three new parameters in VoP2P call-loss models, the first is callee-attendance rate, which is due to the characteristic of destination peer-nodes with a lower attendance probability; the second is call-blocking rate, which is a traditional definition just because the relay-node reaches the limitation of resource; the third is handoff-dropping rate (aka handoff-loss rate), which is caused by the relay-node being removed frequently. The simulation results provide useful reference in the study of teletraffic of VoP2P system.

The rapid growth of Internet population leads to the shortage of IP addresses. The next generation IP protocol, IPv6, which extends the IP address length from 32 bits to 128 bits, was proposed around mid-ninety to accommodate the problem. IP table lookup is based on the longest prefix matching. Most of the existing lookup algorithms scale poorly as IP address space expands to 128 bits. We propose a table lookup scheme for IPv6 using binary search among prefix-length based on parallel processing to improve the performance. We merge hash tables to reduce the lookup complexity of the worst case and use both pipeline and multi-threading to improve the throughput of the average case. Our scheme is realized based on parallel architecture. The simulation results show that the maximum throughput is one lookup per 100 clock cycles in average. This means that, with 600 MHz clock rate, our proposed scheme is able to accomplish 6 million lookups for IPv6 packets. It is more cost effective comparing with existing commercial products in terms of number of lookups per MHz.

Motivated by the promising voice over IP technology, and the wide availability of WLANs, the application of Voice over WLAN (VoWLAN) is expected to encounter dramatic growth in the near future. IEEE 802.11e standard was established to achieve a high level QoS, it introduced a new medium access mechanism HCF in order to solve the QoS provisioning problem in the legacy IEEE 802.11. In this work we propose an adaptive polling scheme, which works on the HC side in HCCA mode, in which HC maintains two dynamic polling lists to reduce both access delay and polling overhead. Both VBR and CBR traffic are taken into consideration. Simulation results showed that the polling overhead is reduced significantly, in addition to high throughout and low access delay comparing to the classical Round-Robin polling scheme and the reference scheme in the standard.

Queue management, bandwidth share and congestion control are very important to both the robustness and fairness of the Internet. A new TCP-friendly router-based active queue management scheme, called WARD, approximates the fair queueing policy. WARD is a simple packet dropping algorithm with a random mechanism and discriminates against the flows which submit more packets per second than is allowed by their fair share. By doing this, it not only protects transmission control protocol (TCP) connections from user datagram protocol (UDP) flows, but also solves the problem of competing bandwidth among different TCP versions, such as TCP Vegas and TCP Reno. Furthermore, it is stateless and easy to implement, so WARD controls unresponsive or misbehaving flows with a minimum overhead. In this article, we present a deterministic fluid model of TCP/WARD system, and explain the UDP throughput behaviour with WARD. Also, we prove that, provided the number of TCP flows is large, the UDP bandwidth share peaks at (2e)−1=0.184 when UDP input rate is slightly larger than link capacity and drops to zero as UDP input rate tends to infinity.

In this article, the problem of providing a fair bandwidth allocation to the flows sharing a congested link in a router is investigated. Queue management, bandwidth share and congestion control are very important to both the robustness and fairness of the Internet. The buffer at the outgoing link is a simple FIFO, shared by packets belonging to the flows. A new transmission control protocol (TCP)-friendly router-based active queue management scheme, termed WARD, is proposed to approximate the fair queueing policy. WARD is a simple packet-dropping algorithm with a random mechanism which discriminates against flows that submit more packets per second than is allowed as their fair share. By doing this, it not only protects TCP connections from user datagram protocol flows, but also solves the problem of competing bandwidth among different TCP versions, such as TCP Vegas and TCP Reno. In addition, WARD works quite well for TCP flow isolation even with different round trip times. In other words, WARD improves the unfair bandwidth allocation properties. Furthermore, as it is stateless and easy to implement, WARD controls unresponsive or misbehaving flows with only a minimum overhead.

In a wireless LAN environment, wireless stations with the strongest received signal can not be guaranteed to have the best quality of service if the population sharing the network capacity was not considered. In other words, within the same access point, the more the population, the less the shared bandwidth, and the worse the quality of service will be. In this paper, we proposed an anticipative agent assistance which is an agent-based metric for evaluating and managing the resource of the wireless access points, computing the potential AP list, and providing clients with resource information of APs. We also propose a novel QoS feedback mechanism which allows users to promptly adjust the service quality with AAA according to the throughput and delay requirements. We evaluate the performance of our proposed method using the ns-2 simulator. Numerical results show that AAA help reduce the transmission delay, increase the throughput, improve the network utilization, accommodate more users, and provide load-balancing

In a wireless network environment, common channel errors, due to multipath fading, shadowing and attenuation, may cause bit errors and packet loss quite different from the packet loss caused by network congestion. In congestion control, the packet loss information can serve as an index of network congestion for effective rate adjustment; therefore, wireless packet loss can mistakenly lead to dramatic performance degradation. The paper proposes a packet loss classification algorithm based on detecting the trend in relative one-way trip time (ROTT) when it falls in the ambiguous zone where the packet loss classification is not straightforward. We show that the proposed algorithm greatly benefits rate-based congestion control algorithms for multimedia over IP networks.

Mobile IP network provides hosts the connectivity to the Internet while changing locations. However, when using TCP Vegas over a mobile network, it may respond to a handoff by invoking a congestion control algorithm, thereby resulting in performance degradation, because TCP Vegas is sensitive to the change of RTT (round-trip time) and it may recognize the increased RTT as a result of network congestion. Since TCP Vegas could not differentiate whether the increased RTT is due to route change or network congestion. This work investigates how to improve the performance of Vegas after a mobile IP handoff and proposes a variation of TCP Vegas, so-called Demo-Vegas, which is able to detect the movement of two end-hosts of a connection, and re-measure the base RTT (minimum round-trip time) if necessary. The proposed mechanism maintains end-to-end semantics, and operates under the existing network infrastructure. Demo-Vegas presents a simple modification in the two end sides of a connection and uses one reserved bit in TCP header. Simulation results demonstrate that Demo-Vegas features higher performance than Vegas in mobile IP networks.

Hashing is a widely used method to perform fast lookup. Several schemes have been proposed to support Internet lookup that includes IP lookup and packet classification. Rectangular search is a well-known packet classification scheme based on multiple hash accesses for different filter length. It shows good scalability with respect to the number of filters; however, the lookup performance is not satisfactory. For example, through experiments, each packet classification takes about 40 hash accesses in a 100,000-filter database and each hash access may take more than one memory access. Obviously, this is insufficient to provide gigabits throughput. We proposed a novel "lookahead caching" which can significantly improve the performance of the hash-based algorithm. The basic idea is to find out the unmatched case for each incoming packet, thus it is different from the traditional caching mechanism. The experimental results indicate that the proposed scheme can improve the performance by a factor of two. The scheme can be further enhanced using parallel processing.

We present a novel solution to the problem of best matching prefix (BMP) which is required in the IP routing lookup. Our approach is based on the trie-based algorithm. The idea is to prune the trie so that the main-searchable portion of the trie can be fitted into the small on-chip SRAM. The algorithm consists of two parts: level smart-compression trie and trie pruning. In the first part, we present the new trie-based algorithm which can provide flexibility, storage efficiency and incremental update. Moreover, the fixed-size data structure eliminates the complexity of the memory management. Secondly, we define the "core" and present how to apply the concept "trie pruning" to achieve fast IP routing lookup. With the currently popular platform, the proposed scheme can provide 40 MPPS without any assumption. While considering route flaps, the performance degrades by only 0.01% with 4000 BGP updates per 30 seconds.

Nowadays, the commonly used table lookup scheme for IP routing is based on the so-called classless interdomain routing (CIDR). With CIDR, routers must find out the best matching prefix (BMP) for IP packet forwarding, which complicates the IP lookup. Since the IP lookup performance is a major design issue for the new generation routers, we investigate the properties of the routing table and present a new IP lookup scheme. By adopting the extreme compression, the size of the forwarding table can be compressed to 360 Kbytes for a large routing table with 58,000 routing entries. It can accomplish one IPv4 route lookup within maximum nine memory accesses. Consequently, the data structure for the incremental update is introduced. With the new data structure, the proposed scheme can accomplish one route update within 500 ns. Even under the circumstance with 4000 route updates per second, the performance degradation is as low as 0.2 %.

The construction of routing tables has been studied extensively. Although existing work has certain advantages, it either uses complicated data structures which result in large storage requirements and high complexity for updating/building the forwarding table, or it is not scalable to fit in IPv6. Lampson et al. (1999) proposed an IP lookup algorithm which performs binary search on prefixes (BSP). The algorithm is attractive, even for IPv6, because of its bounded worst-case memory requirement. For achieving fast forwarding, the cost is the slowing down of insertion. Although this can be justified, the performance of routing-table reconstruction in BGP is too time-consuming to handle frequent route updates. We propose a fast forwarding table construction algorithm, which can handle more than 4,000 route updates per second. Moreover, it is simple enough to fulfil the need of fast packet forwarding. By using the modified multiway search tree, we can further reduce the depth of the tree and eliminate storage for pointers. This reduces the forwarding table size and shortens the lookup time.

A major design issue for the next generation routers is the IP lookup mechanism. The router needs to perform a longest prefix matching on the address lookup for each incoming packet to determine the next hop. Currently, the process is done in software and has become a major performance bottleneck of the router. We propose a fast IP lookup mechanism in which the forwarding table is small enough to fit in an SRAM with very low cost. It also can be implemented in hardware using the pipeline technique. A large routing table with 45,000 routing prefixes can be compressed to a forwarding table with about 430 kbytes in sine by using our proposed method. In the worst case, the number of memory accesses for a lookup is three. When implemented in pipeline technique, the proposed mechanism can achieve one routing lookup per memory access. With current 10 ns SRAM, this mechanism provides approximately 100 million routing lookups per second. Furthermore, the lookup speed can be improved linearly through the speedup of the memory access.