Reliable transmission of high-quality video over ATM networks

The development of broadband networks has led to the possibility of a wide variety of new and improved service offerings. Packetized video is likely to be one of the most significant high-bandwidth users of such networks. The transmission of variable bit-rate (VBR) video offers the potential promise of constant video quality but is generally accompanied by packet loss which significantly diminishes this potential. We study a class of error recovery schemes employing forward error-control (FEC) coding to recover from such losses. In particular, we show that a hybrid error recovery strategy involving the use of active FEC in tandem with simple passive error concealment schemes offers very robust performance even under high packet losses. We discuss two different methods of applying FEC to alleviate the problem of packet loss. The conventional method of applying FEC generally allocates additional bandwidth for channel coding while maintaining a specified average video coding rate. Such an approach suffers performance degradations at high loads since the bandwidth expansion associated with the use of FEC creates additional congestion that negates the potential benefit in using FEC. In contrast, we study a more efficient FEC application technique in our hybrid approach, which allocates bandwidth for channel coding by throttling the source coder rate (i.e., performing higher compression) while maintaining a fixed overall transmission rate. More specifically, we consider the performance of the hybrid approach where the bandwidth to accommodate the FEC overhead is made available by throttling the source coder rate sufficiently so that the overall rate after application of FEC is identical to that of the original unprotected system. We obtain the operational rate-distortion characteristics of such a scheme employing selected FEC codes. In doing so, we demonstrate the robust performance achieved by appropriate use of FEC under moderate-to-high packet losses in comparison to the unprotected system.     

Error resilient transmission of SPIHT coded images over fadingchannels

The paper introduces a method of transmitting error resilient SPIHT coded images over highly error prone Rayleigh fading channels. First, the source significance of the SPIHT coded output is obtained. Based on the significance of the bits, the channel coding is varied accordingly. Channel coding consists of a mixture of rate compatible punctured convolutional (RCPC) codes and interleaving to combat the burst errors produced by the fading channel. An additional error concealment technique is also introduced into the SPIHT decoder to improve its results in cases where errors cause the corruption of the average luminance level during decoding. Comparison with using RS block codes at a total transmission rate of 1.0 bits/pixel is carried out over fading channels with very high error rates to show the superiority of this method over methods using burst error codes     

Quality control for VBR video over ATM networks

Uncontrolled variable-bit-rate (VBR) coded video yields consistent picture quality, but the traffic stream is very bursty. When sent over ATM networks, cell losses may be incurred due to limited buffer capacity at the switches; this could cause severe picture quality degradation. Source rate control can be implemented to generate a controlled VBR bit stream which conforms to specified bit rate bounds and buffer constraints. However, source rate control could result in picture quality degradation too. Hence, for real-time video services, an important issue to address is whether the picture quality degradation incurred by source rate control is within acceptable levels or how to choose the appropriate coding parameters to make it so. We establish quantitatively the relationship between picture quality and source rate control for the case of guaranteed service with different combinations of allocated bandwidth, buffer size, and other key video-coding parameters of MPEG-2. In addition, quality control in the context of two-layered scalable video service (basic and enhanced quality) is also considered. Our study reveals that, in order to maximize both the basic and the enhanced quality, source rate control should be implemented on both layers. The relationships between the two types of quality and different combinations of allocated bandwidths, buffer sizes, and some key coding parameters are also established quantitatively for MPEG-2 SNR scalability     

Bandwidth renegotiation for VBR video over ATM networks

A scheme for delivery or variable bit-rate (VBR) video over asynchronous transfer mode (ATM) networks where bandwidth can be renegotiated during the duration of a call between the video source and the network is considered. Renegotiation can be initiated by either the video source or the network. The video bandwidth requirement is characterized by a usage parameter control (UPC) consisting, in general, of peak rate, burst length, and sustained rate. A baseline design is outlined where rate-control adjusts the source's rate while a new UPC is requested from the network. When granted, the new UPC allows the source to maintain its target quantization and delay requirements. Rate control epochs may be extended when the network blocks UPC requests or sets a lower UPC value to temporally deal with congestion. Simulation results are presented for VBR MPEG video. The results show that with a moderate renegotiation rate the scheme tracks the bandwidth requirements of the source. As a result, the video quality and bandwidth efficiency can be maintained     

Error resilient video coding techniques

We review error resilience techniques for real-time video transport over unreliable networks. Topics covered include an introduction to today's protocol and network environments and their characteristics, encoder error resilience tools, decoder error concealment techniques, as well as techniques that require cooperation between encoder, decoder, and the network. We provide a review of general principles of these techniques as well as specific implementations adopted by the H.263 and MPEG-4 video coding standards. The majority of the article is devoted to the techniques developed for block-based hybrid coders using motion-compensated prediction and transform coding. A separate section covers error resilience techniques for shape coding in MPEG-4     

Error recovery for ATM transmission over wireless channels

The authors present a technique for efficient recovery from transmission errors to be used when transmitting ATM-like data packets in a wireless channel affected by bursty errors (jamming). Issues related to packet format error protection code structure and retransmission protocol are discussed and simulation results are shown that prove the effectiveness of the proposed approach     

Transporting compressed video over ATM networks with explicit-ratefeedback control

We propose a scheme for transmission of variable bit rate (VBR) compressed video for interactive applications using the explicit-rate congestion-control mechanisms proposed for the available bit rate (ABR) service in asynchronous transfer mode networks. Compressed video is inherently bursty, with rate fluctuations over both short and long time scales. This source behavior can be accommodated by the ABR service, since the explicit-rate scheme allows sources to request varying amounts of bandwidth over time. Moreover, when the bandwidth demand cannot be met, the network provides feedback indicating the bandwidth currently available to a connection. In our scheme, the video source rate is matched to the available bandwidth by modifying the quantization level used during compression. We use trace-driven simulations to examine how effective the enhanced explicit-rate scheme is in “rate matching” between the network and the source and the effect on end-to-end delay. We also look at the sensitivity of the proposed scheme to the estimates of the network round-trip times and to inaccuracies in the rate requests made by sources     

Rate adaptive video transmission over ad-hoc networks

A packet control mechanism via a cross-layer feedback to reduce bursts of packet drops for transmission of video over mobile multihop ad-hoc networks is presented. With this approach, the application layer would be capable of controlling the packet transmission flow in accordance with the multihop characteristics of the routing layer.     

An error resilient video coding and transmission solution over error-prone channels

With the rapid development of network and multimedia technology, the error control in video coding and video transmission over error-prone channels has become increasingly important. The DCT based predictive coding and VLC based entropy coding greatly increase the coding efficiency, but make the compressed video stream very sensitive to transmission errors. Therefore, this paper proposes a reliable error resilient video coding and video transmission framework. In order to improve the robustness to packet loss errors, an error resilient video coding algorithm named Z-FMO is proposed. Some channel may bring random bit errors, an adaptive error concealment algorithm based on macroblock boundary gradient namely ECMBG is proposed aiming at such problem. As the indispensable part of a video transmission system, we implement an adaptive video transmission control algorithm JCBAF. Experimental results show that the proposed framework performs well both in R-D performance and subjective quality.     

On the use of frame-based slice size for the robust transmission ofMPEG video over ATM networks

We study the performance requirements for the transport of MPEG video streams over ATM networks. A control scheme is introduced to limit the propagation of impairments through MPEG-based video sequences as a result of cell losses. The scheme assigns different slice sizes to the video frames based on their type and order of appearance within the group of pictures. The main aim of the proposed scheme is to improve the robustness of the video delivery process at a minimum cost. Our results demonstrate the effectiveness of this scheme in reducing the propagation of impairments due to the loss of information during the transmission of MPEG video streams over ATM networks     

Error recovery for interactive video transmission over the Internet

Real-time interactive video transmission in the current Internet has mediocre quality because of high packet loss rates. Loss of packets in a video frame manifests itself not only in the reduced quality of that frame but also in the propagation of that distortion to successive frames. This error propagation problem is inherent in any motion compensation-based video codec. In this paper, we present a new error recovery scheme, called recovery from error spread using continuous updates (RESCU), that effectively alleviates error propagation in the transmission of interactive video. The main benefit of the RESCU scheme is that it allows more time for transport-level recovery such as retransmission and forward error correction to succeed while effectively masking out delays in recovering lost packets without introducing any playout delays, thus making it suitable for interactive video communication. Through simulation and real Internet experiments, we study the effectiveness and limitations of our proposed techniques and compare their performance to that of existing video error recovery techniques including H.263+ (NEWPRED). The study indicates that RESCU is effective in alleviating the error spread problem and can sustain much better video quality with less bit overhead than existing video error recovery techniques under various network environments     

Cell‐loss control in multiplexing of VBR video over ATM networks

In this article we investigate the impacts of interspacing and source polling policies on the cell‐loss rates in transmission of variable bit rate video sources over Asynchronous Transfer Mode networks. We present a mathematical model that finds the approximate optimal starting times of the video sources that minimize the congestion at the multiplexer. Combined with the approximate optimal source starting times, we propose a source ordering and polling policy for reducing the cell‐loss rates. The proposed policy is tested against alternative policies using simulation of pre‐recorded video data. This revised version was published online in June 2006 with corrections to the Cover Date.     

Multiuser cross-layer resource allocation for video transmission over wireless networks

With the advancement of video-compression technology and the wide deployment of wireless networks, there is an increasing demand for wireless video communication services, and many design challenges remain to be overcome. In this article, we discuss how to dynamically allocate resources according to the changing environments and requirements, so as to improve the overall system performance and ensure individual quality of service (QoS). Specifically, we consider two aspects with regard to design issues: cross-layer design, which jointly optimizes resource utilization from the physical layer to the application layer, and multiuser diversity, which explores source and channel heterogeneity for different users. We study how to efficiently transmit multiple video streams, encoded by current and future video codecs, over resource-limited wireless networks such as 3G/4G cellular system and future wireless local/metropolitan area networks (WLANs/WMANs).     

Congestion–distortion optimized video transmission over ad hoc networks

The performance of low-latency video streaming with multipath routing over ad hoc networks is studied. As the available transmission rate of individual links in an ad hoc network is typically limited due to power and bandwidth constraints, a single node transmitting multimedia data may impact the overall network congestion and may therefore need to limit its rate while striving for the highest sustainable video quality. For this purpose, optimal routing algorithms which seek to minimize congestion by optimally distributing traffic over multiple paths are attractive. To predict the end-to-end rate-distortion tradeoff, we develop a model which captures both the impact of encoder quantization and of packet loss due to network congestion on the overall video quality. The validity of the model is confirmed by network simulations performed with different routing algorithms, latency requirements and encoding structures.     

MPEG-2 over ATM for video dial tone networks: issues and strategies

Market growth for PC multimedia and digital video owes largely to the rapid adoption of ISO compression standards by the industry. For video dial tone (VDT) services, the MPEG-2 set of standards have clearly emerged as the preferred coding method for VDT networks. For point-to-point switched video or multimedia connections, ATM has emerged as the technology of choice for switching and transport. This article describes how compressed digital video is transported over a VDT network, what some of the issues are, and how they are being addressed by the industry. It describes a generic VDT reference architecture, and the delivery method of video and multimedia information over such a network     

Transform domain inter-block interleaving schemes for robust image and video transmission in ATM networks

Data interleaving schemes have proven to be an important mechanism in reducing the impact of correlated network errors on image/video transmission. Current interleaving schemes fall into two main categories: (a) schemes that interleave pixel intensity values and (b) schemes that interleave JPEG/MPEG transform blocks. The schemes in the first category suffer in terms of lower compression ratio since highly correlated information in the spatial domain is de-correlated prior to compression. The schemes in the second category interleave DCT transformed blocks. In this case, in the absence of ARQ, if a packet is lost, an entire block may be lost thus yielding poor image quality and making the error concealment task difficult. Interleaving transform coefficients is tricky and error concealment in the presence of lost coefficients is challenging. In this paper, we develop three different interleaving schemes, namely Triangular, Quadrant, and Coefficient, that interleave frequency domain transform coefficients. The transform coefficients within each block are divided into small groups and groups are interleaved with the groups from other blocks in the image, hence they are referred to as inter-block interleaving schemes. The proposed schemes differ in terms of group size. In the Triangular interleaving scheme AC coefficients in each block are divided into two triangles and interleaving is performed among triangles from different blocks. In the Quadrant interleaving scheme, coefficients in each block are divided into four quadrants and quadrants are interleaved. In the Coefficient interleaving scheme, each coefficient in a block is a group and it is interleaved with the coefficients in other blocks. The compression ratio 3 of the proposed interleaving schemes is impressive ranging from 90 to 98% of the JPEG standard compression while providing much higher robustness in the presence of correlated losses. We also propose two new variable end-of-block (VEOB) techniques, one based on the number of AC coefficients per block (VAC-EOB) and the other based on the number of bits per block (VB–EOB). Our proposed interleaving techniques combined with VEOB schemes yield significantly better compression ratios compared to JPEG (2–11%) and MPEG-2 (3–6.7%) standards while at the same time improve the resilience of the coded data in the presence of transmission errors.     

An integrated FEC coding scheme for ATM transmission over regenerative satellite networks

The application of asynchronous transfer mode (ATM) on both wireless and satellite networks requires system adaptation. This adaptation has to improve the overall system's performance, and achieve high quality‐of‐service classes approaching that for fibre‐optic communications. In this paper, a new integrated forward‐error‐correction (FEC) coding scheme is introduced for ATM transmission over regenerative satellite networks. The proposed FEC scheme is a concatenation of two Reed–Solomon codes tailored for the header and payload parts of the ATM cell. This integrated coding scheme is shown to significantly improve the cell loss ratio as compared to the standard CRC code used in the ATM cell header. We obtain both upper and lower performance bounds for the concatenated code and check their accuracy when compared to exact system's performance. Both analytical and simulation results show that a cell loss ratio and bit‐error rate (BER) of 10^?25 and 10^?7 can be, respectively, achieved with minimum delay requirements on the SATCOM link. Finally, an approximation for the system's throughout is obtained. It is shown that using a hybrid selective‐repeat automatic‐repeat‐request (SR‐ARQ) with the RS code, a large throughput of approximately 0.843 can be achieved at BERs lower than 10^?7 for data services. Copyright © 2005 John Wiley & Sons, Ltd.     

IP multicasting over ATM networks

The Internet protocol (IP) multicast model involves a combination of intrasubnet and intersubnet multicast mechanisms. Technologies supporting a given subnet are expected to have native mechanisms for supporting intrasubnet forwarding of packets sent to multicast destinations. Multicast routers attach to subnets and provide intersubnet forwarding of multicast packets, using interdomain multicast routing protocols developed by the Internet Engineering Task Force (IETF). Unfortunately, ATM networks based on UNI 3.0 or UNI 3.1 signaling service do not provide the native multicast support expected by IP. This has led the IETF to develop the “MARS model”-a fairly complex mechanism for emulating intrasubnet multicast support required when running IPs over ATMs. This paper takes a high level look at the IP multicast service, examines the limitations of the ATM point-to-multipoint virtual channel service, and describes the major architectural points of the MARS model