MTCP: scalable TCP-like congestion control for reliable multicast

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.)

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.

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.

New real-time applications frequently involve timing constraints related to accurate services from communication protocols. Concretely, real-time communication protocols utilize timers to implement these constraints between system event occurrences. In this context, the study of congestion control for Internet reliable multicast is at present an active research field related to real-time protocols. In this paper, the authors present an innovative real-time transport protocol named Scalable Reliable Multicast Stair Hybrid (SRMSH) as new hybrid multiple layer mechanism for multicast congestion control providing detection and recovery loss. This work is focused on formal specification of SRMSH approach using Communicating Real-Time State Machines as a formal method. Besides, SRMSH validation is presented within a formal proof framework in order to check the functional safety and liveness properties. As a result, authors outline a dynamical system framework in order to model behavior of their presented solution.

Mobile wireless communication is susceptible to signal blockage, which is loss of signal, typically due to physical obstruction, over a longer duration relative to fading. Measurements indicate that blockage has a significant impact on reliability in both open and rural areas. Reliable multicast, a transport layer mechanism, attempts to gain network performance by eliminating duplicate packets transmitted from a sender to multiple receivers along common paths while providing guaranteed packet delivery to all receivers. The gain achieved by multicasting places limits on the ability to optimize transmission to heterogeneous receiver channel characteristics in an individualized manner at the multicast transport layer. A simple, low overhead protocol extension to mitigate the impact of blockage upon reliable multicast is proposed by piggybacking on reliable multicast congestion control feedback.

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.

We investigate the problem of congestion control for multicast traffic over datagram packet switched networks and present an end-to-end solution to it. The focus of our study is on multimedia collaborative applications. The group members of such applications, typically, span a heterogeneous inter-network, where routers and links may vary widely in their capabilities. Recently, two end-to-end approaches have been introduced for multicast congestion control: hierarchical multicast (a window based approach), and multiple group (a rate based approach). In this paper, we introduce a new end-to-end technique for multicast congestion control that utilizes multiple groups and is window based. We have conducted an analytical study to evaluate our work, which shows encouraging results compared to other techniques.

Improving the performance of the Mobile Ad hoc NETworks (MANETs) is a challenge. MANETs present several constraints such as dynamic topology, wireless link communication and limited resources like bandwidth and energy. Routing without taking into account these constraints degrades performance and aggravates congestion problem. This paper proposes a new solution combining QoS (Quality of Service) routing protocol and flow control mechanism. This QoS routing protocol selects the routes with more resources in an intelligent manner rather than diffusion. It returns the best route offering a higher transmission rate, a less delay and a more stability. This protocol uses a new metric to compute the most stable route. To reinforce the congestion avoidance, we add a flow control mechanism to adjust the sender's transmission rate for each route. The solution is modeled by the ant systems. The results obtained under the Network Simulator (NS2.31) show that our QoS routing protocol improves the network performance compared to QoS-AODV protocol. The addition of flow control mechanism produces significant improvements in network system performance.

Large and sudden variations in packet transmission delays are often unavoidable in wireless networks. Such large delays, refer to as delay spikes (DSs), are likely to exceed several times the typical network round‐trip‐time figures, which can cause TCP spurious timeouts. The spurious timeouts lead to unnecessary retransmissions and reduction of the TCP sender's transmission rate, and degradation of TCP throughput. In this paper we propose a new scheme called DS‐Agent. The spurious timeout is detected by a DS‐Agent and thus TCP sender can response to this spurious timeout accordingly. The simulation results show the better performance of DS‐Agent scheme compared with F‐RTO and TCP Reno in the presence of DSs which is caused by mobility. Copyright © 2006 John Wiley & Sons, Ltd.

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.

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.

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.

Multimedia applications widely use UDP as the transport protocol because UDP can provide a more timely response to packet loss. Unlike TCP, which has a mature built-in window-based congestion control, UDP applications have to construct their own congestion control at the application level. Various protocols have been proposed for such multimedia applications. Packet loss, round-trip delay, and one-way delay are used in these congestion control protocols to adapt to the network dynamics. Most protocols try to achieve TCP-friendliness by using addictive-increase multiplicative decrease (AIMD) algorithm. In this paper, we present a new bandwidth inference congestion control (BIC) protocol, which adapts its rate by non-intrusive bandwidth inference. It relies on one-way delay trend detection to estimate the end-to-end available bandwidth. We also show an effective BIC protocol implementation based on an MPEG-4 FGS video coding system.

Loss-load curves are a recently proposed feedback mechanism for rate-based congestion control in datagram computer networks. In the loss-load model, packet loss inside the network is a direct function of sender transmission rates, and senders have the responsibility of choosing their own transmission rate based on the loss-load tradeoff curve provided by the network. Earlier work has provided the mathematical basis for the loss-load model and provided preliminary simulation results demonstrating its responsiveness, fairness, and stability. The loss-load model works well for simple network environments where each source has many packets to transmit, and wishes to maximize raw throughput.In this paper, the applicability of loss-load curves is extended to senders with more general network traffic requirements, such as file transfer activities. In particular, the paper considers senders wishing to minimize the response time for a transfer of N packets. When N is small, a sender chooses a low transmission rate, to minimize the probability of packet loss. When N is large, a higher transmission rate is chosen to increase raw throughput. The loss-load curve mechanism provides necessary and sufficient information for senders at the edges of the network to choose a proper transmission rate based on their traffic requirements and the current network load.Simulation results show that the loss-load mechanism is effective in achieving high network utilization and minimizing response time for file transfers. The loss-load model also does a good job of providing equitable service to senders with different traffic workloads and different round trip times.

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.