In this paper, we propose a set of broadcast streaming protocols designed for unidirectional radio channels. Considering the limited size and implementation overhead on a mobile terminal, the proposed protocol set is almost compliant with the current mobile streaming protocols, i.e. 3GPP PSS (Packet-switched Streaming Service), except for that the proposed protocols are designed to work on a unidirectional downlink channel. This protocol set enables flexible layout rendering by SMIL (Synchronized Multimedia Integration Language) in combination with SDP (Session Description Protocol), and reliable and synchronized static media (including still image and text) delivery by RTP (Real-time Transport Protocol) carousel. We present the prototype of this protocol set and measure its performance of video quality and waiting time for video presentation through a W-CDMA radio channel emulator and header compression nodes. From the experimental results, we show 1) trade-off between video quality and waiting time, 2) advantage and disadvantage of header compression, 3) effectiveness of synchronized transmission of SDP, SMIL, and I-frames of video objects, and 4) reliability of RTP-carousel. This protocol set is applicable to 3G MBMS (Multimedia Broadcast/Multicast Service) streaming service.

A new inter-client synchronization framework employing a server-client coordinated adaptive playout and error control toward one-to-many (i.e., multicast) media streaming is discussed in this paper. The proposed adaptive playout mechanism controls the playout speed of audio and video by adopting the time-scale modification of audio. Based on the overall synchronization status as well as the buffer occupancy level, the playout speed of each client is manipulated within a perceptually tolerable range. By coordinating the playout speed of each client, the inter-client synchronization with respect to the target presentation time is smoothly achieved. Furthermore, RTCP-compatible signaling between the server and group-clients is performed to achieve the inter-client synchronization and error recovery, where the exchange of controlling message is restricted. Simulation results show the performance of the proposed multicast media streaming framework.

In this paper, a new video broadcast protocol is proposed for video-on-demand (VoD) in shared environment. The new protocol is developed by modifying the first segment delivery scheme for the skyscraper protocol using the idea of patching. The results show that the start-up latency for users is greatly reduced when using our new protocol.

This letter presents the implementation results of an application-level cache file system, MCFS, which is specifically designed to provide efficient caching and transmission mechanisms for streaming media. The file system is built on a virtual file disk which is constructed as a single large file on a general-purpose file system. MCFS suits the access requirement of continuous media caching and provides an efficient I/O mechanism for cache servers. The experimental results show that MCFS outperforms the comparison model and provides a consistent I/O bandwidth.

The proxy mechanism widely used in WWW systems offers low-delay data delivery by means of "proxy server." By applying proxy mechanisms to video streaming system, we expect that high-quality and low-delay video distribution can be accomplished without introducing extra load on the system. In addition, it is effective to adapt the quality of cached video data appropriately in the proxy if user requests are diverse due to heterogeneity in terms of the available bandwidth, end-system performance, and user's preferences on the perceived video quality. In this paper, we propose proxy caching mechanisms to accomplish high-quality and low-delay video streaming services. In our proposed system, a video stream is divided into blocks for efficient use of cache buffer. A proxy cache server is assumed to be able to adjust the quality of cached or retrieved video blocks to requests through video filters. We evaluate our proposed mechanisms in terms of the required buffer size, the play-out delay and the video quality through simulation experiments. Furthermore, to verify the practicality of our mechanisms, we implement our proposed mechanisms on a real system and conducted experiments. Through evaluations from several performance aspects, it is shown that our proposed mechanisms can provide users with a low-latency and high-quality video streaming service in a heterogeneous environment.

In this paper, we present a CDN (Content Delivery Network) architecture for mobile streaming service in which content segmentation, request routing, pre-fetch scheduling, and session handoff are controlled by SMIL (Synchronized Multimedia Integration Language) modification. In this architecture, mobile clients simply follow modified SMIL files downloaded from a portal server; these modifications enable multimedia content to be delivered to the mobile clients from the best surrogates in the CDN. The key components of this architecture are 1) content segmentation with SMIL modification, 2) on-demand rewriting of URLs in SMIL, 3) pre-fetch scheduling based on timing information derived from SMIL, and 4) SMIL updates by SOAP (Simple Object Access Protocol) messaging for session handoffs due to client mobility. This architecture enhances streaming media quality for mobile clients while utilizing network resources efficiently and supporting client mobility in an integrated and practical way. The current status of our prototype on a mobile QoS testbed "MOBIQ" is also reported in this paper.

This letter presents the implementation framework of a video streaming server which uses an optical disk library as a source of media archiving. In order to handle the optical storage subsystem in the framework of disk-based stream service model, we have devised an effective stream scheduling, disk caching, and admission control mechanism. The proposed system has been implemented and its key principles are validated with real experiments.

To support video streaming over the current Internet, a media server usually needs to perform adaptive streaming in combination with congestion control. While existing streaming schemes are mainly designed for particular congestion algorithms and rate shaping techniques, this paper proposes a general buffer-driven adaptive streaming scheme based on control theory. Our scheme can be applied in combination with different congestion control algorithms and different rate shaping (and source coding) techniques. It is also amenable to analysis and easy to implement. Analytical and simulation results show that the proposed scheme gives satisfactory video quality under various Internet conditions while fully utilizing the available network bandwidth.

Unicasting video streams over TCP connections is a challenging problem, because video sources cannot normally adapt to delay and throughput variations of TCP connections. This paper describes a method of extending TCP so that TCP connections can effectively carry hierarchically-encoded layered video streams, while being friendly to other competing connections. We call the method Receiver-based Delay Control (RDC). Under RDC, a TCP connection can slow down its transmission rate to avoid congestion by delaying ACK packet generation at the TCP receiver based on congestion notifications from routers. We present the principle behind RDC, argue that it is TCP-friendly, describe an implementation that uses 1-bit congestion notification from routers, and demonstrate by simulations its effectiveness in streaming hierarchically-encoded layered video.