Multicast Communications in Next-Generation Internet

In recent times, the demand for multicast communications is increasing, both from internet service providers and from the content, media providers or distributors. One of the main obstacles to the wider commercial deployment of internet protocol (IP) multicast is the lack of security for the content being transmitted through IP multicast and communication infrastructure underlying IP multicast. This paper presents a multi-party key computation (MKC) protocol which provides added security to IP multicast. The group key computation, in our protocol, involves multi-party computations using one-time pad and a secret sharing scheme. The proposed protocol is secure, efficient and compatible with IPSec.

Distributed real-time systems call for expressive modeling languages to capture and express their functional and nonfunctional requirements at early stages of the development process. The UML profile for Schedulability, Performance and Time (UML/SPT) is an object-oriented real-time modeling language. UML/SPT has been designed using the built-in extension mechanisms of UML, which makes it flexible and customizable. In this paper, we propose an extension for UML/SPT to capture multicast communications. We define a metamodel that encapsulates the main concepts involved in multicast communications, we show its relationship to UML/SPT domain model, and we introduce new stereotypes corresponding to these concepts. We illustrate the extension with the modeling of the Reliable Multicast Transport Protocol (RMTP2). Finally, we compare our approach to extend UML/SPT for multicast communications with an extension for MSC having the same purpose.

This paper introduces a reliable multicast transport protocol RCCMP (reliable congestion controlled multicast protocol) that has been designed to be simple, scalable, reliable and TCP-friendly. The congestion control is a central part of the protocol, where the feedback of the worst receiver is used to control a transmission window in a TCP-like fashion. The scalability issue is addressed with an exponential timers scheme, that is also used to estimate the number of receivers involved in the communication. The protocol neither needs support from network elements nor maintains state information dependent of the number of receivers. The paper presents a number of simulation results on its performance. Besides, some design choices are evaluated.

IP multicast is advantageous for high quality streaming applications and future needs in the Internet. However, it is generally recognized that IP multicast requires significant routing coordination and configuration, and hence its routing protocols are non-scalable. Recently, Source-Specific Multicast (SSM) has been standardized and proposed as the deployable IP multicast communication architecture. SSM basically works for the one-to-many communication, and eliminates many of the complexities the traditional many-to-many multicast communication has. While SSM gives advantages for the IP multicast deployment, there is still a gap between what is reported as the state-of-the-art in the literature and what could be implemented in practice.In this paper, we analyze the deployment barriers SSM creates, and consider how we can ease some of the barriers. To define the possible approaches, we discuss the functions SSM requires, and the necessary components network operators and application developers need to know for fulfilling the demand.

Network protocols in cellular wireless data networks must update routes as a mobile host moves between cells. These routing updates combined with some associated state changes are called handoffs. Most current handoff schemes in wireless networks result in data loss or large variations in packet delivery times. Unfortunately, many applications, such as real-time multimedia applications and reliable transport protocols, adapt to long term estimates of end-to-end delay and loss. Violations and rapid fluctuations of these estimates caused by handoff processing often result in degraded performance. For example, loss during handoff adversely affects TCP performance [4], and high packet loss and variable delays result in poor real-time multimedia performance. In this paper, we describe a multicast-based protocol that eliminates data loss and incurs negligible delays during a handoff. The basic technique of the algorithm is to anticipate a handoff using wireless network information in the form of received signal strengths and to multicast data destined for the mobile host to nearby base stations in advance. This routing, combined with intelligent buffering techniques at the base stations, enables very rapid routing updates and eliminates data loss without the use of explicit data forwarding. We have implemented this protocol using IP Multicast and Mobile IP-like routing. In our implementation, handoffs typically take between 8 and 15 ms to complete and result in no data loss.

A large number of network services use IP and reliable transport protocols. For applications with constant pressure of data, loss is handled satisfactorily, even if the application is latency sensitive. For applications with data streams consisting of intermittently sent small packets, however, users experience extreme latencies more frequently. Due to the fact that such thin-stream applications are commonly interactive and time-dependent, increased delay may severely reduce the experienced quality of the service. When TCP is used for thin-stream applications, events of highly increased latency are common, caused by the way retransmissions are handled. Other transport protocols that are deployed in the Internet, like SCTP, model their congestion control and reliability on TCP, as do many frameworks that provide reliability on top of unreliable transport. We have tested several application-and transport layer solutions, and based on our findings, we propose sender-side enhancements that reduce the application-layer latency in a manner that is compatible with unmodified receivers running commodity opeating systems like Linux, *BSD, Mac OSX and Windows. We have implemented the mechanisms as modifications to the Linux kernel, both for TCP and SCTP. The mechanisms are dynamically triggered so that they are only active when the kernel identifies the stream as thin. To evaluate the performance of our modifications, we have conducted a wide range of experiments using replayed thin-stream traces captured from real applications as well as artificially generated thin-stream data patterns. From the experiments, effects on latency, redundancy and fairness were evaluated. The analysis of the performed experiments shows great improvements in latency for thin streams when applying the modifications. Surveys where users evaluate their experience of several applications' quality using the modified transport mechanisms confirmed the improvements seen in the statistical analysis. The positive effects of our modifications were shown to be possible without notable effects on fairness for competing streams. We therefore conclude that it is advisable to handle thin streams separately, using our modifications, when transmitting over reliable protocols to reduce retransmission latency.

The efficient support of the IP Integrated Services architecture (IIS) in ATM networks is nowadays a challenging research topic in the internetworking area. The integration of the two architectures needs the overcoming of the inconsistencies existing between their service models, in particular when considering multicast communications. The IIS architecture is based on the Resource Reservation Protocol (RSVP). IP multicast and RSVP rely on a receiver- oriented mode, where a receiver is not only capable of deciding whether to join or leave a multicast sessions, but it is also allowed to choose the associated QoS independently from the other participants. On the other hand, the ATM service model is sender-oriented: the originating node of a multipoint session is the only entity allowed to add or delete leaf nodes. In addition, the originating node sets the QoS of a session, which is unique for all the receivers. The IETF standard for IP multicast over ATM is the MARS model. Nevertheless this approach has been developed for best effort connections only. In this paper we present an extension to the MARS protocol n order to be integrated with the RSVP protocol for supporting QoS management. Within a logical IP Subnet, a Multicast Integration Server (MIS) is entitled of IP multicast address to ATM addressees resolution using the MARS protocol, and of QoS management using RSVP. In addition to the functionalities foreseen by the MARS specification, the MARS server within the MIS uses anew set of messages to inform the sender to a group about the QoS chosen by each receiver. An RSVP entity within the MIS is used to distribute RSVP messages within the LIS and to provide to the MARS the translation of the QoS level form IIS to ATM.© (1998) COPYRIGHT SPIE--The International Society for Optical Engineering. Downloading of the abstract is permitted for personal use only.

This paper introduces a XCP framework based reliable multicast transport protocol XRMCCP (XCP framework based Reliable Multicast Congestion controlled protocol) that has been designed to be simple, scalable and reliable for high bandwidth-delay product (BDP) networks. We observe that TCP Friendly Throughput Equation (TFTE), which many current multicast transport protocols use to estimate available rates of multicast sessions, has limits of performance in high BDP networks. XRMCCP generalizes XCP framework to support one-to-many transport congestion control. The scalability issue is addressed with an exponential timers scheme that is also used to estimate the number of receivers involved in the communication. The paper presents a number of simulation results on its performance. Besides, some design choices are evaluated.KeywordsCongestion ControlRound Trip TimeCongestion WindowFairness IndexBottleneck LinkThese keywords were added by machine and not by the authors. This process is experimental and the keywords may be updated as the learning algorithm improves.

The last few years have seen a great number of new network applications. Many of these applications are characterized by their multipoint features and by their needs of high bandwidth. To allow the deployment of these applications at a large scale in the Internet without running the risk of network implosion, an adapted congestion control mechanism has to be used. In this paper we propose to use the Fair Scheduler paradigm for end-to-end congestion control for the design of a congestion control algorithm to be used with the Reliable Multicast Transport Protocol RMTP. The FS paradigm supposes that all the network routers use a fair scheduler queueing mechanism. This hypotheses allowed us to have worst case bounds on the queueing delay that a block of RMTP data packets experiences between the sender and any receiver in the multicast group. Based on these bounds the sender will regulate its transmission rate in order to avoid a network congestion. Two different versions of this algorithm are presented in detail. The coexistence RMTP/TCP will be the issue of a suite of tests that intend to evaluate the performance of the new algorithm over a Fair Queueing (FQ) scheduled network with the presence of TCP flows sharing the same network resource as the RMTP flow. The same tests will be rerun over a FIFO scheduled network in order to have an idea about the advantages of using the scheduling discipline as a network support for congestion control in the Internet.

New wireless sensor network applications (e.g., military surveillance) require higher reliability than a simple best effort service could provide. Classical reliable transport protocols like Transmission Control Protocol (TCP) are not well suited for wireless sensor networks due to both the characteristics of the network nodes (low computing power, strong energy constraints) and those of the main applications running on those nodes (low data rates). Recent researches present new transport protocols for wireless sensor networks providing various type of reliability and using new mechanisms for loss detection and recovery, and congestion control. This paper presents a survey on reliable transport protocol for WSNs.

This thesis focuses on exploring a multicast communication protocol forWiFi Direct (WD) 802.11 networks. The aim is to improve reliability and efficiency of multicast communication in WD networks. Multicast inWDhas numerous potential applications such as local content sharing, sharing network services, playing multi-player games, and a number of other proximity based services in various fields, viz. health, sports, agriculture, transportation, and gaming. The two major issues with multicast, reliability and efficiency, have been extensively studied in literature for standard WiFi 802.11 networks. However, multicast protocols used in standard WiFi 802.11 networks cannot be straightforwardly used in WD networks because the two technologies are different. Some of these differences include multicast group formation, group architecture, network topology, role and capacity of multicast transmitter. As a result, a multicast protocol cannot achieve reliability and efficiency without taking into account the specifications of the MAC and PHY layers of WD 802.11 networks. Motivated from the leader based approaches of standard WiFi, an Enhanced Leader Based Multicast (ELBM) protocol is proposed to achieve reliability in multicast communication in WD networks. It reduces collision and interference in multicast data transmission by improving the channel access mechanism and selecting an optimal representative multicast receiver. In order to assess the performance improvement which results from early detection of packet collision, Early Packet Loss Detection (EPLD) analytical model is proposed, which surpasses the standard protocol in terms of system throughput. Similarly, a theoretical model is formulated to investigate the new features of the MAC and PHY layers of 802.11ac, such as different Multiple In Multiple Out (MIMO) configurations, Modulation and Coding Schemes (MCS), and wider channel bandwidths under TGn channel models. Because hidden nodes can largely affect the throughput of WD networks, therefore an improved analytical model known as Vidden is developed to analyse the Very High Throughput (VHT) of 802.11ac in the presence of hidden stations. Vidden carefully calculates the collision probability by taking into account both the contending, as well as, the hidden stations A novel adaptive algorithm is proposed to maximize the efficiency of WD 802.11 networks under a TGn channel model by choosing optimal PHY parameters in accordance with targeted Quality of Experience (QoE) for a particular application. The simulation results show that the proposed method outperforms the standard method, thereby achieving an optimal performance in an adaptive manner. Thereafter, an efficient methodology is proposed to reduce the overall Packet Error Rate (PER) based on the simulation results. Finally, the problem of selecting the most favourable transmission channel and rate is investigated for a multicast communication system in the context of WD 802.11 networks. To this end, a novel Multi-rate Multi-channel Multicast (M3-Cast) protocol is proposed, which not only chooses the most favourable communication channel and transmission rate but also takes into account the implementation details of the underlying WD technology, thereby optimizing the overall system performance. M3-Cast is formulated analytically and evaluated by a complete system level simulation. The detailed results and analysis consider a number of performance metrics, such as bit error rate (BER), multicast capacity, and system throughput under different MIMO configurations, channel bandwidths, and various network radii. Consequently, the simulation and analytical results show that M3-Cast protocol outperforms the standard multicast protocol of WD by almost two-fold in terms of system throughput.

Without doubt, multicast communication as a means for one-to-many or many-to-many delivery of data has become a hot topic in multimedia environments. A lot of people are interested in multicast: the research community, standards groups and Internet Service Providers (ISP) among them. Although IP multicast is a very good solution for Internetworking multimedia in many-to-many communications, there are still issues that have not been completely solved. Protocols are still evolving and new protocols are constantly coming up to solve these issues because that is the only way for making multicast become a true Internet service. The main goal of this chapter is to describe the evolution of IP multicast from the obsolete MBone (Multicast Backbone) and intra-domain multicast routing to the actual inter-domain multicast routing scheme. We will pay special attention to the challenges and problems that need to be solved, the problems that have been solved and the way they were solved. We will make a complete picture of the state of the art explaining the idea behind each protocol and how all those protocols work together.

In wireless multimedia sensor networks (WMSNs) a sensor node may have different types of sensor which gather different kinds of data. To support quality of service (QoS) requirements for multimedia applications having a reliable and fair transport protocol is necessary. One of the main objectives of the transport layer in WMSNs is congestion control. We observe that the information provided may have different levels of importance and argue that sensor networks should be willing to spend more resources in disseminating packets carrying more important information. In WMSN event-driven applications, it is critical to report the detected events in the area, resulting in sudden bursts of traffic due to occurrence of spatially-correlated or multiple events, causing loss of data. Also, nodes have very limited power due to hardware constraints. Packet losses and retransmissions resulting from congestion, cost precious energy and shorten the lifetime of sensor nodes. Till now, in WMSNs, Congestion control techniques are based on detection of congestion and recovery, but they cannot eliminate or prevent the occurrence of congestion. Collision is a symptom of congestion in the wireless channel and can result in a time-variant channel capacity. The main intention of this protocol is to be used as a mechanism for reducing congestion in the network by free resources to set accurate rates and priority data needs. If two nodes send their packets in the shortest path to the parent node in a crowded place, a source node must prioritize the data and uses data that have lower priorities of a suitable detour nodes consisting of low or non- active consciously. Proposed algorithm is applied to the nodes near the base station (which convey more traffic) after the congestion detection mechanism detected the congestion. The results obtained from simulation test done by NS-2 simulator indicate that the proposed model is more innovative and presents better performance in compare with CCF, PCCP and DCCP protocols.

Scalability is of paramount importance in the design of reliable multicast transport protocols, and requires careful consideration of a number of problems such as feedback implosion, retransmission scoping, distributed loss recovery, and congestion control. In this article, we present a reliable multicast architecture that invokes active services at strategic locations inside the network to comprehensively address these challenges. Active services provide the ability to quickly and efficiently recover from loss at the point of loss. They also exploit the physical hierarchy for feedback aggregation and effective retransmission scoping with minimal router support. We present two protocols, one for packet loss recovery and another for congestion control, and describe an experimental testbed where these have been implemented. Analytical and experimental results are used to demonstrate that the active services architecture improves resource usage, reduces latency for loss recovery, and provides TCP-friendly congestion control.

A reliable transport protocol with prediction mechanism for urgent information (PMUI) in WSNs is proposed. With PMUI, which is based on RTP-UI, the congestion control mechanism is improved, and the priority control and prediction mechanisms are adopted by taking the current queue length with change rate and expected queue length with remaining length together into consideration. The congestion status of current queue is analyzed and the changing trend of the next cycle of the queue is predicted as well. In order to evaluate the degree of congestion, state machine is adopted. Working states of WSNs are classified into eight states in accordance with different degrees of congestion. According to the change rate of a queue and the expected change rate of a queue, different rate adjustment mechanisms and bandwidth allocation schemes based on different working states are developed. Simulation results show that PMUI is lower than RTP-UI in both packet loss rate and the average delay for reliable transmission of urgent information.