Abstract not found

The increasing diversity of Internet application requirements has spurred recent interest in transport protocols with flexible transmission controls. In window-based congestion control schemes, increase rules determine how to probe available bandwidth, whereas decrease rules determine how to back off when losses due to congestion are detected. The control rules are parameterized so as to ensure that the resulting protocol is TCP-friendly in terms of the relationship between throughput and loss rate. This paper presents a comprehensive study of a new spectrum of window-based congestion controls, which are TCP-friendly as well as TCP-compatible under RED.

Abstract—The next-generation wireless Internet (NGWI) is expected to provide a wide range of services including high-speed data and real-time multimedia to mobile users. To realize this expectation, a diverse set of challenges need to be addressed, which are posed by heterogeneous wireless networking environments within NGWI and the according application requirements. Furthermore, the architectural heterogeneities must be captured dynamically, while mobile users may roam during their connection duration. Current existing transport layer protocols have been developed for a specific network paradigm in mind, e.g., for wireless local area networks (WLANs), micro/macro wireless systems, or for satellite systems. Using these existing different transport layer protocols for NGWI to support global roaming of mobile users is not a practical solution due to processing and memory constraints of wireless terminals. Thus, there is a need

Abstract—Next-generation wireless Internet (NGWI) is expected to provide a wide range of services including real-time multimedia to mobile users. However, the real-time multimedia traffic transport requires rate control deployment to protect shared Internet from unfairness and further congestion collapse. The transmission rate control method must also achieve high throughput and satisfy multimedia requirements such as delay or jitter bound. However, the existing solutions are mostly for the wired Internet, and hence, they do not address the challenges in the wireless environments which are characterized by high bit error rates. In this paper, a new analytical rate control (ARC) protocol for real-time multimedia traffic over wireless networks is presented. It is intended to achieve high throughput and multimedia support for real-time traffic flows while preserving fairness to the TCP sources sharing the same wired link resources. Based on the end-to-end path model, the desired behavior of a TCP source over lossy links is captured via renewal theory. The resulting asymptotic throughput equation is designated as the driving equation for the proposed rate control method. Performance evaluation via simulation experiments reveals that ARC achieves high throughput and meets multimedia traffic expectations without violating good citizenship rules for the shared Internet. Index Terms—Equation-based rate control, jitter bound, realtime multimedia, wireless networks. I.

Abstract—To provide ubiquitous access to the proliferating rich media on the Internet, scalable streaming servers must be able to provide differentiated services to various client requests. Recent advances of transcoding technology make network-I/O bandwidth usages at the server communication ports controllable by request schedulers on the fly. In this article, we propose a transcodingenabled bandwidth allocation scheme for service differentiation on streaming servers. It aims to deliver high bit rate streams to high priority request classes without overcompromising low priority request classes. We investigate the problem of providing differentiated streaming services at application level in two aspects: stream bandwidth allocation and request scheduling. We formulate the bandwidth allocation problem as an optimization of a harmonic utility function of the stream quality factors and derive the optimal streaming bit rates for requests of different classes under various server load conditions. We prove that the optimal allocation, referred to as harmonic proportional allocation, not only maximizes the system utility function, but also guarantees proportional fair sharing between classes with different prespecified differentiation weights. We evaluate the allocation scheme, in combination with two popular request scheduling approaches, via extensive simulations and compare it with an absolute differentiation strategy and a proportionalshare strategy tailored from relative differentiation in networking. Simulation results show that the harmonic proportional allocation scheme can meet the objective of relative differentiation in both short and long timescales and greatly enhance the service availability and maintain low queueing delay when the streaming system is highly loaded. Index Terms—Service differentiation, harmonic proportional bandwidth allocation, video transcoding, streaming bit rate, feedback queue. 1

Layered (LC) and multiple description coding (MDC) have been proposed as source coding techniques that are robust to channel errors for video transmission. LC and MDC have similar characteristics: they both generate multiple sub-bitstreams, and it is permissible to drop some portion of the data from the sub-bitstreams during transmission for both methods. However, they are different in the sense that the sub-bitstreams for LC have different levels of importance while all sub-bitstreams for MDC are equally important. Since these two encoding techniques have similar properties, some performance comparisons between LC and MDC have recently been reported. However, these studies are still not conclusive because several scenarios have not been carefully considered. Furthermore, they have been performed in different environments. In this paper, we further investigate the error-resilience capabilities of these two encoding techniques through extensive experimentation. Although some of our conclusions agree with those in the literature, we believe that this paper provides the most comprehensive performance comparison yet between LC and MDC.

In this paper we present a video communication system that integrates end-to-end buffer management and congestion control at the source with the playout adjustment mechanism at the receiver. While each component of the system has been considered independently in the literature, our focus in this work is their integration. The proposed system exploits the fact that when congestion control is implemented at the source, most of the loss occurs at the source and not within the network. Based on this observation, we design the buffer management to trade off random loss for controlled loss of visually less important data. Frame rate is adjusted at the receiver to maximize the visual quality of the displayed video based on the overall loss. We tested our system with both H.26L and a subband/wavelet video coder, and found that it significantly improves the received video quality in both cases.

Abstract — We consider live unicast video streaming over a packet erasure channel. To protect the transmitted data, previous solutions use forward error correction (FEC), where the channel code rate is fixed in advance according to an estimation of the packet loss rate. However, these solutions are inefficient under dynamic and unpredictable channel conditions because of the mismatch between the estimated packet loss rate and the actual one. We introduce a new approach based on rateless codes and receiver feedback. For every source block, the sender keeps on transmitting the encoded symbols until it receives an acknowledgment from the receiver indicating that the block was decoded successfully. Within this framework, we provide an efficient algorithm to minimize bandwidth usage while ensuring successful decoding subject to an upper bound on the packet loss rate. Experimental results showed that compared to traditional fixed-rate FEC, our scheme provides significant bandwidth savings for the same playback quality. I.

This paper describes the ARMS system which enables secure and adaptive rich media streaming to a large-scale, heterogeneous client population. The secure streaming algorithms ensure end-to-end security while the content is adapted and streamed via intermediate, potentially untrusted servers. ARMS streaming is completely standards compliant and to our knowledge is the first such end-to-end MPEG-4-based system.

Video application over the Internet are getting increasingly popular because of the explosive growth of the Internet. However, video packets loss due to network congestions can degrade the video quality substantially. In this paper, we propose a transmission scheme for Motion-JPEG2000 video sequences over IP networks. Our scheme utilizes the progression modes in Motion-JPEG2000. It can be implemented in an active network environment efficiently. Our simulation shows that our scheme gracefully adapts to network congestion and improves the quality of video transmission in congested IP networks.