Receiver-driven Layered Multicast Research Articles

Prioritization-based Layered Multicast for Fixed/Mobile Networks with Fast Convergence and Inter-Session Fairness

Multimedia streaming services are becoming popular in both wired and wireless networks. Layered multicast is a widely accepted approach for streaming multimedia data to a large number of users. Existing layered multicast approaches donot interact well with network dynamics. Indeed, upon a change in network conditions, they require a long time till they can appropriately adjust their data transmission rate. Additionally, they do not achieve fairness when users from different sessions share the bandwidth of a bottleneck link.In this paper, we propose a scheme that allows newly-arriving users to promptly converge their data transmission rates to the most optimal rate that best suit the current conditions of the network without degrading the system fairness. The proposed scheme is based on the fact that layered multicast uses prioritybased packet dropping policies. In the proposed scheme, two newly-defined packet messages are considered: “low priority join” and “normal join” messages. To join a session, a user first subscribes to all corresponding layers by issuing “low priority join” messages. It then computes packet drops experienced oneach layer. If packets of a given layer experience a drop rate higher than a predetermined threshold, the user leaves that layer and all higher layers. The user then “officially” joins the remaining layers by transmitting “normal join” messages. This operation helps users to subscrive to only layers whose aggregate bandwidth fits the current network conditions. The performance of the proposed scheme is evaluated through computer simulations and is compared against the Receiverdriven Layered Multicast (RLM) scheme. The results show that the proposed scheme achieves appropriate bandwidth utilization from the start of the session. The results demonstrate also that the proposed scheme is effective in managing handover in mobile networks and achieves better Quality of Service (QoS) inheterogeneous mobile environments.

Robust and fair Multicast Congestion Control (M2C)

Since 1995 and the Receiver-driven Layered Multicast (RLM) protocol, numerous multicast congestion control protocols have been proposed, such as RLC, FLID-SL, FLID-DL and finally the WEBRC protocol. However these protocols suffer from some limitations and have not been extensively experimented on a wide range of network conditions. This article presents the Multicast Congestion Control (M2C) protocol, which is based on a dynamic layer mechanism and a congestion window mechanism for each receiver. M2C is designed to be TCP-friendly and to fairly share the bandwidth when competing with unicast and multicast streams.To reduce simulation approximations and to carry out extensive evaluations over the Internet we implemented M2C and several other multicast congestion control protocols. The experiments reveal interesting network behaviors, such as the join time latency problem, that simulations and local testbeds had not revealed before. We evaluated the impact of this latency on the state of the art protocols. Focusing on WEBRC and M2C, we have then defined and run a validation benchmark to qualify their behavior in terms of fairness and convergence time. These experiments show M2C robustness and good behavior for the mentioned metrics and its improvements compared to WEBRC. Furthermore, we have analyzed the signaling overhead of these protocols.Thus, M2C is an efficient multicast congestion control with an open source implementation which is easily usable by multicast applications, such as file transfer or video streaming.

Wyner–Ziv Video Compression and Fountain Codes for Receiver-Driven Layered Multicast

The increasing popularity of video streaming applications that distribute data to a large number of clients motivates the design of reliable multimedia delivery systems capable of adapting to diverse transmission conditions. Receiver-driven layered multicast (RLM) efficiently addresses the issue of heterogeneity in clients' available bandwidths and packet loss rates by shifting rate control to the receiver side. We propose a system for RLM over the Internet and 3G wireless networks based on layered Wyner-Ziv video coding and digital fountain codes. Layered Wyner-Ziv video coding improves robustness to packet loss compared to current scalable video coders, such as MPEG-4 FGS coder, while generating a scalable output bit stream. Digital fountain codes are near-capacity erasure protection codes that are ideally suited for multicast applications due to their rateless property. By combining an error-resilient Wyner-Ziv video coder and rateless fountain codes, our system allows reliable video multicast to an arbitrary number of heterogeneous receivers without the requirement of feedback channels. Simulation results show performance improvements over a previous scheme that exploits multiple description and layered coding.

Discrete bandwidth allocation considering fairness and transmission load in multicast networks

As a promising solution to tackle the network heterogeneity in multicasting, layered multicast protocols such as receiver-driven layered multicast (RLM) and layered video multicast with retransmission (LVMR) have been proposed. This paper considers fairness as well as transmission load in the layered multicasting. Lexicographically fair bandwidth allocation among multicast receivers is considered under the constraint of minimum bandwidth requirement and the link capacity of the network. The problem of transmission load in the layer multicasting due to various user requirements is also examined by minimizing the number of layers. The bandwidth allocation is formulated as a nonlinear integer programming problem. A dual-objective tabu search is proposed to solve the fairness and transmission load problem in multicast networks. Outstanding performance is obtained by the proposed tabu search. When the fairness objective is considered, the solution gap from the optimal solution is less than 2% in problems with 50 virtual sessions. The complexity of the dual objective largely depends on the weighting factor of the two objectives. Even in tough cases the proposed tabu search provides excellent solution, whose gap is within 6% from the optimal solution.

Fine-grained Layered multicast with STAIR

Traditional approaches to receiver-driven layered multicast have advocated the benefits of cumulative layering, which can enable coarse-grained congestion control that complies with TCP-friendliness equations over large time scales. In this paper, we quantify the costs and benefits of using noncumulative layering and present a new, scalable multicast congestion control scheme called STAIR that embodies this approach. Our first main contribution is a set of performance criteria on which we base a comparative evaluation of layered multicast schemes. In contrast to the conventional wisdom, we demonstrate that fine-grained rate adjustment can be achieved with only modest increases in the number of layers, aggregate bandwidth consumption and control traffic. The STAIR protocol that we subsequently define and evaluate is a multiple rate congestion control scheme that provides a fine-grained approximation to the behavior of TCP additive increase/multiplicative decrease (AIMD) on a per-receiver basis.

Sender-adaptive and receiver-driven layered multicast for scalable video over the Internet

In this paper, we propose and analyze a new system architecture for video multicast over Internet, namely, the sender-adaptive and receiver-driven layered multicast (SARLM). In SARLM, the sender of a video source splits the video data coded by a scalable codec and a channel codec into multiple data streams, each of which corresponds to a separate multicast group. The sender can adjust the way in which the video sequence is split dynamically based on the receivers' network parameters collected through feedback. Meanwhile, a receiver can estimate available bandwidth based on a modified packet-pair technique and choose to reassemble and playback the video sequence for a given quality level by dynamically subscribing a given part or all of the data streams according to its network conditions. To optimize the sender's adaptation strategy, we introduce a quality-space (Q-Space) model to describe and analyze the mathematical relationship between the sending rate of different SARLM layers and the video quality received by a given receiver identified by its network characteristics including available bandwidth and packet loss ratio. Our simulation results demonstrate that, under the same network topology and condition, the SARLM architecture can achieve higher network throughput and better video qualities on the receiver side than the existing approaches.

Combined Wavelet Video Coding and Error Control for Internet Streaming and Multicast

This paper proposes an integrated approach to Internet video streaming and multicast (e.g., receiver-driven layered multicast (RLM) by McCanne) based on combined wavelet video coding and error control. We design a packetized wavelet video (PWV) coder to facilitate its integration with error control. The PWV coder produces packetized layered bitstreams that are independent among layers while being embedded within each layer. Thus, a lost packet only renders the following packets in the same layer useless. Based on the PWV coder, we search for a multilayered error-control strategy that optimally trades off source and channel coding for each layer under a given transmission rate to mitigate the effects of packet loss. While both the PWV coder and the error-control strategy are new--the former incorporates embedded wavelet video coding and packetization and the latter extends the single-layered approach for RLM by Chou et al.--the main distinction of this paper lies in the seamless integration of the two parts. Theoretical analysis shows a gain of up to 1 dB on a channel with 20% packet loss using our combined approach over separate designs of the source coder and the error-control mechanism. This is also substantiated by our simulations with a gain of up to 0.6 dB. In addition, our simulations show a gain of up to 2.2 dB over previous results reported by Chou et al.

Achieving inter-session fairness for layered video multicast

The Internet is increasingly used to deliver multimedia services. Since there are heterogeneous receivers and changing network conditions, it has been proposed to use adaptive rate control techniques such as layered video multicast to adjust the video traffic according to the available Internet resources. A problem of layered video multicast is that it is unable to provide fair bandwidth sharing between competing video sessions. We propose two schemes, layered video multicast with congestion sensitivity and adaptive join-timer (LVMCA) and layered video multicast with priority dropping (LVMPD), to achieve inter-session fairness for layered video multicast. Receiver-driven layered multicast (RLM), layer-based congestion sensitivity, LVMCA, and LVMPD are simulated and compared. Results show both proposed schemes, especially LVMPD, are fairer and have shorter convergence time than the other two schemes.

Revisiting the fair queuing paradigm for end-to-end congestion control

Today, the dominant paradigm for congestion control in the Internet is based on the notion of TCP friendliness. To be TCP-friendly, a source must behave in such a way as to achieve a bandwidth that is similar to the bandwidth obtained by a TCP flow that would observe the same round-trip time (RTT) and the same loss rate. However, with the success of the Internet comes the deployment of an increasing number of applications that do not use TCP as a transport protocol. These applications can often improve their own performance by not being TCP-friendly, which severely penalizes TCP flows. To design new applications to be TCP-friendly is often a difficult task. The idea of the fair queuing (FQ) paradigm as a means to improve congestion control was first introduced by Keshav (1991). While Keshav made a fundamental step toward a new paradigm for the design of congestion control protocols, he did not formalize his results so that his findings could be extended for the design of new congestion control protocols. We make this step and formally define the FQ paradigm as a paradigm for the design of new end-to-end congestion control protocols. This paradigm relies on FQ scheduling with per-flow scheduling and longest queue drop buffer management in each router. We assume only selfish and noncollaborative end users. Our main contribution is the formal statement of the congestion control problem as a whole, which enables us to demonstrate the validity of the FQ paradigm. We also demonstrate that the FQ paradigm does not adversely impact the throughput of TCP flows and explain how to apply the FQ paradigm for the design of new congestion control protocols. As a pragmatic validation of the FQ paradigm, we discuss a new multicast congestion control protocol called packet pair receiver-driven layered multicast (PLM).

NLM: network-based layered multicast for traffic control of heterogeneous network

Layered multicast has been considered to be an efficient approach for coping with the network heterogeneity. Receiver-driven Layered Multicast (RLM) and its variants suggest that every receiver adjusts the reception level based on the packet loss rate. In this paper we propose a scheme called network-based layered multicast (NLM) scheme, which enhances the packet scoping mechanism employed in current multicast backbone (MBone). It makes use of the TTL bits and TOS bits in IP header, and packets are filtered in the routers using both the measured and predicted traffic condition. This allows the network quickly and efficiently adapts to the traffic condition. We evaluate and compare the scheme with the receiver-driven schemes using Network Simulator for a 33-node network topology with realistic bandwidths. The simulation reveals that the proposed NLM scheme substantially improves the stability and fairness of the RLM scheme for multicast sessions, while it displays comparable performance to the RLM with Priority (RLMP) scheme. For mixed type traffic of multicast and TCP sessions, it outperforms both the schemes.

Error control for receiver-driven layered multicast of audio and video

We consider the problem of error control for receiver-driven layered multicast of audio and video over the Internet. The sender injects into the network multiple source layers and multiple channel coding (parity) layers, some of which are delayed relative to the source, Each receiver subscribes to the number of source layers and the number of parity layers that optimizes the receiver's quality for its available bandwidth and packet loss probability. We augment this layered FEC system with layered pseudo-ARQ. Although feedback is normally problematic in broadcast situations, ARQ can be simulated by having the receivers subscribe and unsubscribe to the delayed parity layers to receive missing information. This pseudo-ARQ scheme avoids an implosion of repeat requests at the sender and is scalable to an unlimited number of receivers, We show gains of 4-18 dB on channels with 20% loss over systems without error control and additional gains of 1-13 dB when FEC is augmented by pseudo-ARQ in a hybrid system, Optimal error control in the hybrid system is achieved by an optimal policy for a Markov decision process.

Low-complexity video coding for receiver-driven layered multicast

The Multicast Backbone, or MBone, has risen from a small, research curiosity to a large-scale and widely used communications infrastructure. A driving force behind this growth was the development of multipoint audio, video, and shared whiteboard conferencing applications. Because these real-time media are transmitted at a uniform rate to all of the receivers in the network, a source must either run at the bottleneck rate or overload portions of its multicast distribution tree. We overcome this limitation by moving the burden of rate adaptation from the source to the receivers with a scheme we call receiver-driven layered multicast, or RLM. In RLM, a source distributes a hierarchical signal by striping the different layers across multiple multicast groups, and receivers adjust their reception rate by simply joining and leaving multicast groups. We describe a layered video compression algorithm which, when combined with RLM, provides a comprehensive solution for scalable multicast video transmission in heterogeneous networks. In addition to a layered representation, our coder has low complexity (admitting an efficient software implementation) and high loss resilience (admitting robust operation in loosely controlled environments like the Internet). Even with these constraints, our hybrid DCT/wavelet-based coder exhibits good compression performance. It outperforms all publicly available Internet video codecs while maintaining comparable run-time performance. We have implemented our coder in a real application-the UCB/LBL videoconferencing tool vic. Unlike previous work on layered video compression and transmission, we have built a fully operational system that is currently being deployed on a very large scale over the MBone.

Receiver-driven layered multicast

State of the art, real-time, rate-adaptive, multimedia applications adjust their transmission rate to match the available network capacity. Unfortunately, this source-based rate-adaptation performs poorly in a heterogeneous multicast environment because there is no single target rate --- the conflicting bandwidth requirements of all receivers cannot be simultaneously satisfied with one transmission rate. If the burden of rate-adaption is moved from the source to the receivers, heterogeneity is accommodated. One approach to receiver-driven adaptation is to combine a layered source coding algorithm with a layered transmission system. By selectively forwarding subsets of layers at constrained network links, each user receives the best quality signal that the network can deliver. We and others have proposed that selective-forwarding be carried out using multiple IP-Multicast groups where each receiver specifies its level of subscription by joining a subset of the groups. In this paper, we extend the multiple group framework with a rate-adaptation protocol called Receiver-driven Layered Multicast, or RLM. Under RLM, multicast receivers adapt to both the static heterogeneity of link bandwidths as well as dynamic variations in network capacity (i.e., congestion). We describe the RLM protocol and evaluate its performance with a preliminary simulation study that characterizes user-perceived quality by assessing loss rates over multiple time scales. For the configurations we simulated, RLM results in good throughput with transient short-term loss rates on the order of a few percent and long-term loss rates on the order of one percent. Finally, we discuss our implementation of a software-based Internet video codec and its integration with RLM.