Multicast congestion control for multimedia collaborative applications in packet switched networks

We present MTCT, a congestion control scheme for large-scale reliable multicast. Congestion control for reliable multicast is important because of its wide applications in multimedia and collaborative computing, yet nontrivial, because of the potentially large number of receivers involved. Many schemes have been proposed to handle the recovery of lost packets in a scalable manner; but there is little work on the design and implementation of congestion control schemes for reliable multicast. We propose new techniques that can effectively handle instances of congestion occurring simultaneously at various parts of a multicast tree. Our protocol incorporates several novel features: (1) hierarchical congestion status reports that distribute the load of processing feedback from all receivers across the multicast group, (2) the relative time delay (RTD) concept which overcomes the difficulty of estimating round-trip times in tree-based multicast environments, (3) window-based control that prevents the sender from transmitting faster than packets leave the bottleneck link an the multicast path through which the sender's traffic flows, (4) a retransmission window that regulates the flow of repair packets to prevent local recovery from causing congestion, and (5) a selective acknowledgment scheme that prevents independent (i.e., non-congestion-related) packet loss from reducing the sender's transmission rate. We have implemented MTCP both on UDP in SunOS 5.6 and on the simulator ns, and we have conducted extensive Internet experiments and simulation to test the scalability and inter-fairness properties of the protocol. The encouraging results we have obtained support our confidence that TCP-like congestion control for large-scale reliable multicast is within our grasp.

Information-centric networking (ICN) is expected to be a candidate for future internet architecture, and it supports features such as multicast that improves bandwidth utilization and transmission efficiency. However, multicast itself does not provide congestion control. When multiple multicast groups coexist, multicast traffic may exhaust all network resources, and cause network congestion and packet loss. Additionally, traditional IP multicast congestion control mechanisms cannot be directly applied to ICN architecture. Therefore, it is necessary to consider an effective congestion control mechanism for ICN multicast. This paper proposes a single-rate multicast congestion control mechanism, called SRMCC. It supports router-assisted awareness of the network congestion state and congestion control message aggregation. Moreover, the fair shared rate estimation method is innovatively proposed to achieve protocol fairness. Most importantly, it adjusts the rate according to different congestion states indicated by the queue occupancy ratio. By introducing a rate selection factor, it can achieve a balance between packet loss rate and throughput. Experimental results show that our proposal outperforms other mechanisms in throughput, packet loss rate, total bandwidth utilization, and overhead, and achieves protocol fairness and better TCP friendliness.

In this article we present a simulation-based comparison of one of the best-known multicast congestion control schemes—TCP-friendly Multicast Congestion Control (TFMCC)—against our proposed Adaptive Smooth Multicast Protocol (ASMP). ASMP consists of a single-rate multicast congestion control mechanism which takes advantage of the RTCP Sender (SR) and Receiver Reports (RR) in order to adjust the sender's transmission rate in respect of the network conditions. The innovation in ASMP lays in the ‘smooth’ transmission rate, which is TCP-friendly and prevents oscillations. We use an integrated simulation environment named Multi-Evalvid-RA for the evaluation of the two congestion control schemes. Multi-Evalvid-RA provides all the necessary tools to perform simulation studies and assess video quality by using both network-centric metrics along with video quality measurements. Performance evaluation results show that ASMP is a very efficient solution for rate-adaptive multimedia applications and a serious competitor to well-known TFMCC. Copyright © 2012 John Wiley & Sons, Ltd. (Uncontrolled video transmission without any flow/congestion control mechanisms should be avoided as it leads to poor-quality service due to packet losses. Therefore, moderate transmission rates that satisfy the video application requirements are preferable to avoid packet losses. Otherwise, error resilience mechanisms should be applied to recover from losses, especially from I-frames that cannot be recovered by the video decoders. More efficient video encoding techniques along with error resilience and higher access capacity are the key elements for increasing video quality.)

Maintaining the performance of reliable transport protocols, such as transmission control protocol (TCP), over wireless mesh networks (WMNs) is a challenging problem due to the unique characteristics of data transmission over WMNs. The unique characteristics include multi-hop communication over lossy and non-deterministic wireless mediums, data transmission in the absence of a base station, similar traffic patterns over neighboring mesh nodes, etc. One of the reasons for the poor performance of conventional TCP variants over WMNs is that the congestion control mechanisms in conventional TCP variants do not explicitly account for these unique characteristics. To address this problem, this paper proposes a novel artificial intelligence based congestion control technique for reliable data transfer over WMNs. The synergy with artificial intelligence is established by exploiting a carefully designed neural network (NN) in the congestion control mechanism. We analyze the proposed NN based congestion control technique in detail and incorporate it into TCP to create a new variant that we name as intelligent TCP or iTCP. We evaluate the performance of iTCP using both ns-2 simulations and real testbed experiments. Our evaluation results demonstrate that our proposed congestion control technique exhibits a significant improvement in total network throughput and average energy consumption per transmitted bit compared to the congestion control techniques used in other TCP variants.

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.

End-to-end multicast congestion control (MCC) is a complex problem. There exist two categories of multicast congestion control. The first category is single rate MCC, where the source adjusts its rate according to the slowest receiver within the session. The second category is multi-rate MCC, where the source has to disseminate the data in multiple layers and each receiver could subscribe to the required layers according to their own capacities. Among the recently proposed multicast congestion control protocols is TFMCC, which is an equation based single rate protocol. In TFMCC, at each RTT, each receiver estimates its throughput using an equation that estimates the steady-state throughput of a TCP source. The receiver with the lowest such value is declared as the current limiting receiver (CLR). The source then adjusts its rate to that of the CLR. Our objections to TFMCC are with respect to the equation adopted as well as to the slow reaction of the sender to congestion conditions. In this paper we propose two modifications to TFMCC. First, each receiver estimates its throughput using an equation that models the steady state throughput of a multicast source controlled according the additive increase multiplicative decrease (AIMD) approach. The second modification consists of monitoring RTT variations at the source side in order to rapidly adjust the sending rate to network conditions. Simulation results show that these two modifications results in remarkable performance improvement with respect to throughput, time to react, and magnitude of oscillations.

Congestion in Wireless Multimedia Sensor Networks (WMSN) may cause packet loss, buffer overflow and increases queueing delay, which has a significant negative impact on the network performance and reliability. This problem imposes more resource constraints that involve energy consumption and buffer size to meet additional requirements such as desired buffer size and limited amount of packet loss. One of the fundamental issues of congestion problem in such networks is the queue management. In fact, handling congestion without considering flexible buffer management and queueing methods, stay insignificant against dropping rate and packet loss. Consequently, a novel congestion control approach is needed to overcome these problems. In this paper, a new queue management mechanism called: Priority Queue Management based Explicit Congestion Control (PQMECC) is proposed and modeled using Stochastic Automata Networks, which are appropriate for modeling and analyzing complex systems with interacting components and complex synchronization. The congestion control technique used in PQMECC is based on priority, load repartition mechanism and explicit congestion notification which is generated only if the congestion reaches the threshold value.

Congestion control is an active research field and numerous experimental protocols have been proposed. This paper presents a panorama of various congestion avoidance and control techniques developed for unicast internet transport protocols. We summarize the congestion control in TCP. Next, we present a synthesis of improvements relating to acknowledgments management in one hand, and congestion window size regulation in the other hand. Then, we summarize main works on congestion-window-less protocols, and the impact on transport protocol of routers information. Perspectives and concluding remarks end the paper.

A new Dynamic Allocation bandwidth strategy for Multicast Congestion Control(DAMCC) was put forward.Concerning the current problems of adjusting the rate of granularity roughly and inadequate application about the Receiver-driven Layered Congestion control(RLC),this strategy designed an algorithm,which can calculate the rate of the enhancement layer dynamically and separately.The receiver of the strategy of DAMCC can calculate the value of the Round-Trip Time(RTT) according to the feedback response information,then calculate its own TCP friendly rate and receive the multicast data at the corresponding rate to achieve sharing the network resources with TCP flow fairly.The simulation results show DAMCC strategy can use network bandwidth more effectively than the RLC strategy,and it also solves the problem of network bandwidth heterogeneity.Besides,it can calculate the TCP friendly rate at the receiver to achieve sharing the network resources with the TCP flow fairly.

Congestion control for large multicast groups has been a challenging issue for widespread deployment of multicast services. Single-rate multicast congestion control schemes address both TCP-friendly and scalability issues with low complexity. However, they do not solve the well-known drop-to-zero problem, i.e., to prevent a slow receiver from slowing down faster receivers in the same multicast group. For this purpose, we propose a new single-rate multicast congestion control mechanism for heterogeneous receiver which helps each multicast receiver to receive at its desired rate. The basic idea is as follows. Each multicast receiver maintains and adjusts a window at itself just like what TCP source does. They calculate a desirable receive rate with this window and the RTT to the source and feed back this rate to the source and intermediate routers. The sender tunes its rate according to the fastest receiver in the group. Each multicast router does not copy each incoming multicast packet to all egress links in the multicast tree unconditionally but copies packets to egress links according to the rate information obtained from the downstream receivers. Good egress links receive more packets while bad egress links receive less packets. The proposed strategy is shown to be still TCP-friendly and scalable while eliminating the drop-to-zero problem.

Multicast is attracting a resurgence of interest because it has a potential to address the explosively growing need for efficient streaming of large-volume Internet content. However, to realize the potential, large-scale multicast distribution requires effective congestion control. In this paper, we design SIM, a protocol that combines three distinct mechanisms (Selective participation, Intra-group transmission adjustment, and Menu adaptation) to provide a general solution for efficient fair scalable privacy-preserving multicast congestion control with heterogeneous receivers. Whereas the individual mechanisms have appeared in earlier multicast protocols, our main contribution lies in the cohesive integration of the techniques. SIM achieves such integration by operating the three mechanisms at different timescales and distributing the responsibility for control to different entities in the network. Besides, to instantiate and integrate the three control mechanisms, SIM employs several novel techniques of independent interest. Our extensive evaluation confirms the salient properties of SIM in diverse settings where receiving capabilities are highly heterogeneous, bottleneck capacities fluctuate, bottlenecks migrate, and session membership is dynamic.

The study of congestion control for Internet reliable multicast is at present an active research field. In this paper we present scalable reliable multicast stair hybrid (SRMSH) as a multiple layer mechanism for multicast congestion control providing detection and recovery loss. We simulated SRMSH with different traffic sources to measure heterogeneity, scalability and fairness. Our work focus on bandwidth analysis and results clearly conclude that SRMSH exhibits reasonable throughput using these metrics and a good performance competing with typical traffic flow from Internet like TCP.

The main essence of this proposed method for the control of congestion in VANET is exploiting the already existing resources of the network and at the same time preventing the unnecessary overheads of nodes as well as links of a network. The control of congestion in the networks such as vehicular ad hoc suffers through a special kind of challenges. These challenges are the consequences of the environmental specificities such as frequently changing of the topology, large amount of variations in the nodes density, nature of the communication in broadcasting and the configurational characteristics of the neighbor etc. Keeping above described point into mind, we proposed a technique for the control of unnecessary congestion by reducing the number of transmitting messages in VANET. It supports discarding of similar types of messages. Here each node consists of a neighbor table for comparing the incoming messages with the previous messages. The information from the message is extracted and the values are compared with the values of neighbor table using a database query. If a match is found then the corresponding message field in the table is updated else a new entry is added in the table and the message is transferred to control queue (CQ) or the service queue (SQ).After being processed the entry is deleted from the queue as well as table. The working efficiency of this developed method is determined on the ground of the average delay of these three main types of messages v/s its mean message size.

This paper presents a evaluation on existing multicast congestion control schemes. We evaluate several recently proposed schemes to summarize state of the art in this field. Solutions to typical problems in multicast congestion control, especially those in layered multicast, such as layer granularity, loss path multiplicity and receiver coordination, are also studied to motivate further research.

In mobile ad-hoc networks, the multicast paradigm is of central importance. It can help to save scarce medium bandwidth if packets are to be delivered to multiple destinations. We consider the problem of congestion control for multicast traffic in wireless multihop networks. We propose to apply a congestion control concept which is tailored to the very special properties of the wireless multihop medium: implicit hop-by-hop congestion control. The idea, so far only having been considered for unicast traffic, is here generalized to multicast. We implement it in the Backpressure Multicast Congestion Control (BMCC) protocol, with a focus on how to realize it in combination with geographic multicast routing in the Scalable Position-Based Multicast (SPBM) protocol. Our evaluation points out a number of highly desirable properties of the proposed scheme. In particular, it achieves and maintains high throughput and high packet delivery ratios at low packet latencies, even in the presence of significant network load.