This letter proposes an efficient Two-stage Resource scheduling algorithm for cloud based Live Media Streaming system (TRLMS). It transforms the cloud-based resource scheduling problem to a min-cost flow problem in a graph, and solves it by an improved Successive Short Path (SSP) algorithm. Simulation results show that TRLMS can enhance user demand satisfaction by 17.1% than mean-based method, and its time complexity is much lower than original SSP algorithm.

For real-time video streaming, tree-based Application Level Multicasts (ALMs) are effective with respect to transmission delay and jitter. In particular, multiple-tree ALMs can alleviate the inefficient use of upload bandwidth among the nodes. However, most conventional multiple-tree ALMs are constructed using a Distributed Hash Table (DHT). This causes considerable delay and consumes substantial network resources because the DHT, generally, does not take distances in the IP network into account. In addition, the network constructed by a DHT has poor churn resilience because the network needs to reconstruct all the substreams of the tree network. In this paper, we propose a construction method involving overlapped cluster trees for delivering streamed data that are churn resilient. In addition, these overlapped cluster trees can decrease both the delay and the consumption of network resources because the node-connecting process takes IP network distances into account. In the proposed method, clusters are divided or merged using their numbers of members to optimize cluster size. We evaluated the performance of the proposed method via extensive computer simulations. The results show that the proposed method is more effective than conventional multiple-tree ALMs.

This letter proposes a new delivery format in order to realize unified transmissions of stereoscopic video contents over a dynamic adaptive streaming scheme. With the proposed delivery format, various forms of stereoscopic video contents regardless of their encoding and composition types can be delivered over the current dynamic adaptive streaming scheme. In addition, the proposed delivery format supports dynamic and efficient switching between 2D and 3D sequences in an interoperable manner for both 2D and 3D digital devices, regardless of their capabilities. This letter describes the designed delivery format and shows dynamic interoperable applications for 2D and 3D mixed contents with the implemented system in order to verify its features and efficiency.

In recent years, real-time streaming has become widespread as a major service on the Internet. However, real-time streaming has a strict playback deadline. Application level multicasts using multiple distribution trees, which are known as forests, are an effective approach for reducing delay and jitter. However, the failure or departure of nodes during forest-based multicast transfer can severely affect the performance of other nodes. Thus, the multimedia data quality is degraded until the distribution trees are repaired. This means that increasing the speed of recovery from isolation is very important, especially in real-time streaming services. In this paper, we propose three methods for resolving this problem. The first method is a random-based proactive method that achieves rapid recovery from isolation and gives efficient “Randomized Forwarding” via cooperation among distribution trees. Each node forwards the data it receives to child nodes in its tree, and then, the node randomly transferring it to other trees with a predetermined probability. The second method is a reactive method, which provides a reliable isolation recovery method with low overheads. In this method, an isolated node requests “Continuous Forwarding” from other nodes if it detects a problem with a parent node. Forwarding to the nearest nodes in the IP network ensures that this method is efficient. The third method is a hybrid method that combines these two methods to achieve further performance improvements. We evaluated the performances of these proposed methods using computer simulations. The simulation results demonstrated that our proposed methods delivered isolation recovery and that the hybrid method was the most suitable for real-time streaming.

This letter proposes a new mechanism that supports adaptive sending control using Real-Time Streaming Protocol (RTSP) and Transmission Control Protocol (TCP) for IPTV service over heterogeneous networks. The proposed mechanism is implemented on a mobile IPTV device and its performance is verified for providing seamless television watching in heterogeneous networks, even when in motion.

With the fast development of mobile communication technologies, mobile multimedia services like mobile Video on Demand (VOD) are becoming prevalent. However, VOD streaming requires dedicated bandwidth to satisfy Quality of Service (QoS), and the limited wireless bandwidth will become insufficient to support the increasing number of mobile VOD users. In order to solve the problem, this paper proposes a Call Admission Control (CAC) approach which can accept new users even when the system bandwidth is insufficient. Our approach also guarantees continuous playback for subscribers by taking into account the service end time and the delay bound of the users. Simulation results demonstrate that the proposed scheme can increase the number of concurrent users and reduce the connection blocking probability significantly without playback interruption.

This paper deals with two types of capacity allocation schemes, i.e., static and adaptive, for uplink and downlink burst durations in the IEEE 802.16 BE (Best Effort) service. We study QoE (Quality of Experience) enhancement of audio-video IP transmission over the uplink channel with the two capacity allocation schemes. We introduce a piggyback request mechanism for uplink bandwidth requests from subscriber stations to the base station in addition to a random access-based request mechanism. We assess QoE of audio-video streams for four schemes obtained from the combination of the capacity allocation schemes and the bandwidth request mechanisms. We also employ two types of audio-video contents. From the assessment result, we notice that the adaptive allocation scheme is effective for QoE enhancement particularly under heavily loaded conditions because of its efficient usage of OFDM symbols. In addition, the piggyback request mechanism can enhance QoE of audio-video transmission. We also find that the effects of capacity allocation schemes and piggyback request mechanism on QoE change according to the content types.

In this paper, we propose an adaptive chunk scheduling for mesh-based peer-to-peer live streaming system, a hybrid class of push and pull chunk delivery approach. The proposed rule-based push-pull scheduler simultaneously pull video chunk from lower latency peers to fill up missing chunks and push video chunk adaptively for rapid chunk delivery. We performed comparative simulation study against rarest first push-pull and status-wise push-pull to prove the efficiency of our proposed algorithm. Mesh-push is made possible by effectively exploiting the information through buffer map exchange. The findings of performance evaluation have suggested a better video continuity and achieved lower source to end delay.

Establishing peer-to-peer (P2P) live streaming for mobile ad hoc network (MANET) requires an efficient scheme to deliver the real-time data in the infrastructure-less disaster environment. However, P2P membership management is difficult in the dynamic mobility and resource limited MANET. In this paper, we present a cross-layer design for P2P-MANET which integrates P2P DHT-based routing protocol and IPv6 routing protocol. Therefore, the proposed scheme can manage and recover the P2P overlay as well as selecting efficient routing path to multicast video streaming. The simulation results demonstrate that the proposed scheme performs relatively better than the layered approach or the off-the-shelf design in terms of the playback continuity and signaling overhead.

In the infrastructure-less disaster environment, the application of the peer-to-peer (P2P) group conference over mobile ad hoc network (MANET) can be used to communicate with each other when the rescue crews search the survivors but work separately. However, there still are several problems of in-time multimedia delivery in P2P-MANET: (1) MANET mobility influences the maintenance of P2P overlay. (2) P2P overlay is not proximal to MANET topology, this leads to the inefficient streaming delivery. (3) The unreliable wireless connection leads to the difficulty of multi-source P2P group conferencing. Therefore, P2P conferencing cannot work well on MANET. To overcome the above disadvantages, in this paper, we present a cross-layer P2P group conferencing mechanism over MANET, called RING (Real-time Intercommunication Network Gossip). The RING uses the ring overlay to manage peers and utilizes the cross-layer mechanism to force the ring overlay to be proximal to MANET topology. Therefore, RING can lead efficient in-time multimedia streaming delivery. On the other hand, the ring overlay can deal with peer joining/leaving fast and simply, and improves the delivery efficiency with the minimum signaling overhead. Through mathematical theory and a series of experiments, we demonstrate that RING is workable and it can shorten the source-to-end delay with minimal signaling overhead.

Most large-scale Peer-to-Peer (P2P) live streaming systems are constructed as a mesh structure, which can provide robustness in the dynamic P2P environment. The pull scheduling algorithm is widely used in this mesh structure, which degrades the performance of the entire system. Recently, network coding was introduced in mesh P2P streaming systems to improve the performance, which makes the push strategy feasible. One of the most famous scheduling algorithms based on network coding is R2, with a random push strategy. Although R2 has achieved some success, the push scheduling strategy still lacks a theoretical model and optimal solution. In this paper, we propose a novel optimal pull-push scheduling algorithm based on network coding, which consists of two stages: the initial pull stage and the push stage. The main contributions of this paper are: 1) we put forward a theoretical analysis model that considers the scarcity and timeliness of segments; 2) we formulate the push scheduling problem to be a global optimization problem and decompose it into local optimization problems on individual peers; 3) we introduce some rules to transform the local optimization problem into a classical min-cost optimization problem for solving it; 4) We combine the pull strategy with the push strategy and systematically realize our scheduling algorithm. Simulation results demonstrate that decode delay, decode ratio and redundant fraction of the P2P streaming system with our algorithm can be significantly improved, without losing throughput and increasing overhead.

Nowadays portable devices with multiple wireless interfaces and using multimedia services are becoming more popular on the Internet. This paper describes a family of multipath binomial congestion control algorithms for audio/video streaming, where a low variant of transmission rate is important. We extend the fluid model of binomial algorithms for single-path transmission to support the concurrent transmission of packets across multiple paths. We focus on the extension of two particular algorithms, SQRT and IIAD, for multiple paths, called MPSQRT and MPIIAD, respectively. Additionally, we apply the design technique (using the multipath fluid model) for multipath TCP (MPTCP) into the extension of SQRT and IIAD, called fbMPSQRT and fbMPIIAD, respectively. Both two approaches ensure that multipath binomial congestion control algorithms achieve load-balancing, throughput improvement, and fairness to single-path binomial algorithms at shared bottlenecks. Through the simulations and comparison with the uncoordinated protocols MPSQRT/MPIIAD, fbMPSQRT/fbMPIIAD and MPTCP, we find that our extended multipath transport protocols can preserve lower latency and transmission rate variance than MPTCP, fairly share with single-path SQRT/IIAD, MPTCP and TCP, and also can achieve throughput improvements and load-balancing equivalent to those of MPTCP under various scenarios and network conditions.

The peer-assisted streaming paradigm has been widely employed to distribute live video data on the internet recently. In general, the mesh-based pull approach is more robust and efficient than the tree-based push approach. However, pull protocol brings about longer streaming delay, which is caused by the handshaking process of advertising buffer map message, sending request message and scheduling of the data block. In this paper, we propose a new approach, mesh-push, to address this issue. Different from the traditional pull approach, mesh-push implements block scheduling algorithm at sender side, where the block transmission is initiated by the sender rather than by the receiver. We first formulate the optimal upload bandwidth utilization problem, then present the mesh-push approach, in which a token protocol is designed to avoid block redundancy; a min-cost flow model is employed to derive the optimal scheduling for the push peer; and a push peer selection algorithm is introduced to reduce control overhead. Finally, we evaluate mesh-push through simulation, the results of which show mesh-push outperforms the pull scheduling in streaming delay, and achieves comparable delivery ratio at the same time.

Recently, video streaming services using P2P (Peer-to-Peer) have attracted attention to solve the problem of load concentration on servers and to reduce large latency. Many P2P streaming systems, like Coolstreaming, however, take a complicated approach to control playback timing severely. This leads to less churn resiliency and less adaptability to fluctuation of network traffic. Therefore, we focus on a simple and robust approach to realize “pseudo” streaming with high quality, which is based on BitTorrent. In the existing methods with the simple approach, peers download pieces just closer to playback timing to decrease the playback discontinuity. However, these methods do not consider the constitution of video structure in sophisticated manner. P2P streaming system must consider several important metrics for high-quality and fair distribution. Therefore, in this paper, we propose a new P2P video streaming system considering the cooperation of three important metrics; video structure, playback timing, and piece dispersion on network. In this system, users vary three piece selections to suit the delivery status. Specifically, users preferentially download pieces which affect the video quality, which are closer to playback timing, and which improve the delivery efficiency. Moreover, we show the effectiveness of the proposed method by computer simulation.

This paper presents a novel chunk selection strategy for peer-to-peer video streaming, called enr-first selection policy, which simultaneously considers both block rarity and playback deadline. The policy intends to boost overall network-wide streaming performance, while ensuring good playback continuity of individual peers. The efficacy of the proposed scheme is validated through our performance evaluation study that demonstrates a substantial gain.

This paper presents novel hybrid push-pull protocols for peer-to-peer video streaming. Our approaches intend to reap the best of push- and pull-based schemes by adaptively switching back and forth between the two modes according to video chunk distributions. The efficacy of the proposed protocols is validated through an evaluation study that demonstrates substantial performance gains.

Since the Content Distribution Network (CDN) and IP multicast have heavy infrastructure requirements, their deployment is quite restricted. In contrast, peer-to-peer (P2P) streaming applications are independent on infrastructures and thus have been widely deployed. Emerging wireless ad-hoc networks are poised to enable a variety of streaming applications. However, many potential problems, that are trivial in wired networks, will emerge when deploying existing P2P streaming applications directly into wireless ad-hoc networks. In this paper, we propose a goodput optimization framework for P2P streaming over wireless ad-hoc networks. A two-level buffer architecture is proposed to reassign the naive streaming systems' data requests. The framework adopts a chunk size-varying transmission algorithm to obtain smooth playback experience and acceptable overhead and utilize limited bandwidth resources efficiently. The distinguishing features of our implementation are as follows: first, the framework works as a middleware and is independent on the streaming service properties; existing P2P streaming application can be deployed in wireless ad-hoc networks with minimum modifications and development cost; second, the proposed algorithm can reduce unnecessary communication overheads compared with traditional algorithms which gain high playback continuity with small chunk size; finally, our scheme can utilize low bandwidth transmission paths rather than discarding them, and thus improve overall performance of the wireless network. We also present a set of experiments to show the effectiveness of the proposed mechanism.

Per-flow traffic measurement is essential for network management; billing, traffic engineering, mitigating denial of service attacks, to mention just a few. In this field, the fundamental problem is that the size of expensive SRAM is too small to hold traffic data from high-speed networks. In this paper, we propose a new method for per-flow traffic measurement, which is based on the virtual vector that was originally designed for the problem of spread estimation. We modify the original virtual vector and show that this simple change yields a highly effective per-flow traffic estimator. Experiments show that our proposed scheme outperforms the state-of-the-art method in terms of both processing time and space requirement.

This paper defines a Quality of Experience (QoE) parameter for multi-view video streaming service over Wireless Broadband Internet, and proposes cross-layer optimization (CLO) algorithm to maximize the QoE. The proposed CLO algorithm interacts with all layers from PHY layer to video layer. Under the time-varying wireless channel condition, it guarantees service continuity to the user by selecting frames to be transmitted. The CLO technique takes inter-view and the temporal significance of the multi-view video frames into consideration when deciding the Transmission Frame Set (TFS) and its Levels of FEC (LOF). Through simulation of the proposed technique, it is shown that the optimal quality of the multi-view video can be achieved even during hand-over, using the minimum amount of resources. Visual quality of multi-view video streams is improved by about 15.330.2%, especially up to 39.165% during the hand-over. Moreover, continuity of the most important view has been increased by 20.139.1% in comparison with conventional method.

Stream Control Transmission Protocol (SCTP) is a new transport layer protocol for the next generation Internet. SCTP is a connection-oriented protocol that carries over TCP's features but also supports UDP-like message-oriented data transmission. In this paper, we make use of SCTP's multi-streaming feature to transmit MPEG-4 video efficiently, and evaluate its transmission performance under the policy with/without differentiated retransmission. Moreover, to enhance the communication quality, we extend SCTP multi-streaming to realize selective retransmission policy. Our extension utilizes packet-by-packet timestamps to control retransmission of lost packets. By computer simulation, we show that SCTP can (1) improve the video quality by exploiting the multi-streaming and partial reliability features, (2) enhance the video transmission quality by adjusting SCTP fast retransmit threshold, and (3) SCTP with our selective retransmission extension can further improve the whole performance.