Experimental study on wireless multicast scalability using Merged Hybrid ARQ with staggered adaptive FEC

We report the implementation experience and experimental evaluation of a staggered adaptive forward error correction (FEC) system for video multicast over wireless LANs. In the system, the parity packets generated by a cross-packet FEC code are transmitted at a time delay from the original video packets, i.e. staggercasting video stream and FEC stream in different multicast groups. The delay provides temporal diversity to improve the robustness of video multicast, especially to enable the clients to correct burst packet loss using FEC and to achieve seamless handoff. A wireless client dynamically joins the FEC multicast groups based upon its channel conditions and handoff events. We have implemented the system including the streaming server and client proxy. A novel software architecture is designed to integrate the FEC functionality in the clients without requirement for changing the existing video player software. We conduct extensive experiments to investigate the impact of FEC overhead and the delay between the video stream and FEC stream to the video quality under different interference levels and mobile handoff durations. The efficacy of staggered adaptive FEC system on improving video multicast quality is demonstrated in real system implementation.

In this paper, we propose a data-driven joint modulator and demodulator based on meta-learning for hybrid automatic repeat request (HARQ) schemes, which include Type-I HARQ, HARQ with chase combining (HARQ-CC), and HARQ with incremental redundancy (HARQ-IR). Data-driven deep learning (DL) techniques have proven particularly effective for enabling wireless systems to adapt to complex scenarios with unknown or imperfect channel conditions that are difficult to characterize analytically. Most existing DL approaches to realize data-driven communication frameworks are based on joint source-channel coding (JSCC). However, the difficulty in detecting transmission errors makes JSCC unsuitable for HARQ systems. To address this limitation, we propose a novel architecture that combines classical channel coding with DLbased modulation and demodulation. Specifically, we employ an autoencoder to implement both the modulator and demodulator. To quickly adapt to time-varying fading channels, the ModelAgnostic Meta-Learning (MAML) algorithm is leveraged to accelerate the convergence of the model as well as enhance the reliability of HARQ. The MAML framework is divided into two phases, including the pre-training phase to acquire shared initialization parameters across various channel conditions and the fine-tuning phase to rapidly adapt to specific channel states. Simulation results demonstrate that the proposed architecture offers dramatic improvements in convergence speed and generalization capabilities, and meanwhile boosts the reliability of HARQ schemes.

There are two kinds of technique used for error control: FEC (forward error correction) and ARQ (automatic repeat request). Combining these two schemes, which is called HARQ (hybrid ARQ), makes it possible to increase the channel throughput. The error of the channel quality estimate that is used to select the MCS (modulation coding scheme) in type I HARQ can be a serious problem in scenarios when the mobile station moves at high speeds. Furthermore, in a system dominated by packet services, the interference level may be bursty, and thus it is difficult to predict the interference level. Therefore, the type II HARQ scheme is a more robust solution compared with type I HARQ. In order to improve channel throughput while at the same time satisfying quality of service (QoS) under various channel conditions, an efficient hybrid ARQ type II scheme is proposed. The optimum redundancy size for every transmission is calculated to maximize the throughput using QoS parameters, the performance of LDPC code and channel conditions. The proposed scheme, together with such benefits as QoS support, provides a way of achieving higher throughput in wireless communication systems.

Named Data Networking (NDN) is considered as a prominent architecture towards future Wireless Local Area Networks (WLAN), and multicast plays an important role in data delivery such as media streaming, multipoint videoconferencing, etc. However, to achieve high-efficiency multicast in NDN WLAN is challenging for two significant reasons. First, without feedback mechanism in IEEE 802.11 standards, to guarantee reliability, the current multicast scheme transmits the multicast data with the basic rate (e.g., 1 Mbps for IEEE 802.11b), which inevitably increases the transmission delay for high-speed consumers. Second, as a NDN multicast group is constituted by consumers who are requesting the same content, multicast groups are easy to form and evolve rapidly, where a data rate adaptation scheme is requisite to accommodate differential multicast groups. In this paper, we propose a multi-stage multicast rate adaptation scheme for NDN WLAN, named <italic xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">NDN-MMRA</i> , to minimize the total transmission time with reliability guarantee for multicast group members. In <italic xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">NDN-MMRA</i> , by checking the Pending Interest Table (PIT) status information, the number of consumers in each multicast group as well as their receiving capabilities are known ahead; with the available data rates in a specific 802.11 standard, <italic xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">NDN-MMRA</i> determines: 1) how many transmission stages are required; and 2) in each stage, which data rate should be adopted. The merit is that with multi-stage transmissions, the data rate can be adapted in descending order to accommodate high-speed consumers with delay minimized, and low-speed consumers with reliability guaranteed. We implement <italic xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">NDN-MMRA</i> in NS-3 by adopting the ndnSIM module, and conduct extensive experiments to demonstrate its efficacy under different IEEE 802.11 standards and various underlying WLAN topologies.

In this paper, we propose a cross-layer design which combines adaptive modulation and coding (AMC) at the physical layer with a hybrid automatic repeat request (HARQ) protocol at the data link layer to minimize the packet error rate (PER) and maximize the spectral efficiency. By using Markov model to predict the channel condition of next frame according to the second order statistics characteristics of the former frame, we calculate the conditional probability density function and joint probability density function of signal to noise ratio (SNR). Further, we derive the packet error rate and average spectral efficiency in closed-form for transmission over Nakagami-m fading channels. Numerical results show that the proposed scheme can obtain lower PER and higher spectral efficiency c ompared with traditional HARQ protocol s. The cross-layer design scheme combing type-III HARQ with AMC can achieve the highest spectral efficiency and the lowest PER than type-I HARQ and type-II HARQ.

In this article, we propose video delivery schemes insuring around 1s delivery latency with Dynamic Adaptive Streaming over HTTP (DASH), which is a standard version of HTTP Live Streaming (HLS), so as to benefit from the video representation switching between successive video segments. We also propose HTTP/2-based algorithms to apply video frame discarding policies inside a video segment when a selected DASH representation does not match with the available network resources. The current solutions with small buffer suffer from rebuffering events. Rebuffering not only impacts the Quality of Experience (QoE) but also increases the delivery delay between the displayed and the original video streams. In this work, we completely eliminate rebuffering events by developing optimal and practical video frame discarding algorithms to meet the 1s latency constraint. In all our algorithms, we request the video frames individually through HTTP/2 multiple streams, and we selectively drop the least meaningful video frames thanks to HTTP/2 stream resetting feature. Our simulations show that the proposed algorithms eliminate rebuffering while insuring an acceptable video quality with at least a Peak Signal to Noise Ratio (PSNR) of 35dB compared to 25dB of the basic First In First Out (FIFO) algorithm. We also quantify and qualify the resulting temporal distortion of the video segments per algorithm. An important number of missing video frames results in a temporal fluidity break known as video jitter . The displayed video looks like a series of snapshots. We show that both the optimal Integer Linear Program (ILP) and practical algorithms decrease the frequency and duration of the jitters. For example, practical algorithms reduce the number of crashed displayed videos (presenting one jitter longer than 1,350ms) with 22% compared to the basic FIFO algorithm. We also show that requesting video frames separately with HTTP/2 slightly increases the overhead from 4.34% to 5.76%.

In this paper, we propose a modified MAC protocol supporting real-time video traffic over the IEEE 802.11 WLAN. Investigating H.26L real-time video transmission over wireless IEEE 802.11b LANs, we present a novel hybrid automatic repeat request (ARQ) algorithm that efficiently combines forward error control (EEC) coding with the ARQ protocol based on multiple transmission stages. The scheme includes: (1) multiple transmission stages with corresponding ARQ feedback information that determines how many transmission stages should be activated; (2) conditional frame skipping algorithm based on accumulated feedback information. Through analysis and simulation, we show that multiple-stages based Hybrid ARQ scheme has high error recovery rate for real-time video frames. In case of severe unreliable channel which causes losing a lot of IP packets, at the cost of little end-to-end delay's increase, we obtain a peak signal-to-noise ratio improvement up to about 10dB compared to the hybrid ARQ proposed in I A. Majumdar et al. (2002).

This paper presents a cross-layer design of adaptive modulation and coding (AMC) with truncated hybrid automatic repeat request (HARQ) for one-to-many multicast transmission, in order to increase the spectral efficiency under the quality-of-service (QoS) constraints. For adaptive transmission over the channel common to the multicast group users, the AMC mode is chosen with the minimum SNR among the users to guarantee the target performance of all users. In the meanwhile, the minimum SNR required to support an AMC mode is aggressively designed by allowing retransmission with HARQ schemes. For the proposed design, we derive the average packet error rate, average number of transmission, and spectral efficiency and provide the performance numerically obtained. Numerical results show that the cross-layer design for multicast provides a significant performance gain at a small number of retransmissions as in unicast. In particular, it is observed that AMC design with HARQ is more beneficial in the low SNR region where multicast performs worse than unicast.

The accessibility of good smartphones in 4G is encouraging mobile users to use video services. This huge increase in the usage of video communication over wireless networks poses many challenges according to fluctuations of wireless channel conditions along with user mobility. The challenge in providing video multicasting over wireless networks would most likely be the channel condition of each user in the same multicast group which is probably not identical due to location of the user and/or congestion of the network. In this paper, an adapted multicasting scheme has been proposed, which gracefully adapts the video quality to heterogeneous receivers and varying network conditions. Results obtained from simulation indicate that an enhancement of video quality under numerous channel conditions was through graceful degradation of these channels in terms of average throughput and packet loss ratio.

Video multicast over Wireless Wide Area Networks (WWAN) is difficult because of unavoidable packet losses and impracticality of retransmission on a per packet, per client basis, due to the known NAK implosion problem. Recent approach exploits clients' cooperation for packet recovery, so that a peer group's received WWAN packets are shared using a secondary network like Wireless Local Area Network (WLAN). For multiview video multicast, where a client can switch views interactively by subscribing to different WWAN multicast channels streaming different views, two new difficulties arise. First, system must provide timely view-switching mechanism, so that client can switch to a desired view quickly for correct decoding and display. Second, it is difficult for system to leverage neighboring peers for cooperative loss recovery, since neighbors are more likely to be subscribing to different views than a loss-stricken peer. In this paper, we use Distributed Source Coding (DSC), a new compression tool in video coding, to solve both problems. Each DSC frame is encoded with a set of predictor frames, and correct decoding only requires one of the predictors in the set to be available at decoder. Periodic insertion of DSC frames into video streams then enables a peer to switch from view v to v' at the DSC frame boundary, assuming DSC frame of view v' was encoded using a frame in view v as one predictor. For the same assumption, a neighbor watching view v can help a peer watching view v' evade error propagation resulting from earlier losses and resume decoding at the DSC boundary. Experiments show that optimized usage of DSC frames in a coding structure, where unequal error protection is enabled to decrease the probability of decoding failure earlier in a group of pictures, outperforms a structure using I-frames instead for view switching by up to 11dB in video quality in typical WWAN network loss environment.

The digital set-top box (STB) has evolved to provide not only broadcast channel videos to TV watchers but also streaming videos to mobile device users. The users connected to a STB may request the same video content with disparate bitrate requirements. In this case, applying the conventional multicast or unicast technique causes a waste of transmission bandwidth. To resolve this problem, in this study we realized a novel hybrid multicast and unicast video streaming system. The unicast service was based on MPEG-DASH (Dynamic Adaptive Streaming over Hypertext Transfer Protocol). For multirate streamed users, we incorporated the H.264 SVC (Scalable Video Coding) for bitrate adaptation into the multicast groups under the FLUTE (File Delivery over Unidirectional Transport) protocol. Furthermore, we optimized the system performance that intelligently determined the transmitted video segment and quality level according to the bandwidth efficiency and the marginal quality benefit to users. Simulation results showed that the proposed algorithm effectively reduced the waiting time and enhanced the received video quality of mobile device users.

In this paper, we propose a cross-layer design (CLD) scheme combining rate adaptation and four types of hybrid automatic repeat request (HARQ) for multiband orthogonal frequency division multiplexing (MB-OFDM) ultra wideband (UWB) systems following the ECMA-368 standard. The time varying property is incorporated into standard UWB channel models for the purpose of investigating rate adaptation. To accurately accommodate fast time-varying channel conditions, we propose to embed the selected rate information into the acknowledgement (ACK) frames. It is shown that the proposed CLD scheme combining rate adaptation and HARQ can provide higher throughput than the HARQ-only schemes. Among the four types of HARQ schemes, HARQ Type-III has the best throughput performance while Type I has the worst.

Multiple multicast groups for multimedia on the Internet

The paper compares two hybrid automatic repeat request (HARQ) schemes using incremental redundancy (IR) which have been proposed for the UMTS (W-CDMA) high speed downlink shared channel (HS-DSCH). Their relative throughput performance is reported for various channel conditions. It is shown that the two-stage rate-matching scheme has marginally worse performance compared to the alternative block interleaving scheme for certain coding rates and under some channel conditions.

With the proliferation of wireless multimedia applications, multicast/broadcast has been recognized as an efficient technique to transmit a large volume of data to multiple mobile stations at the same time. In most multicast systems, the transmitter (e.g. base station) adapts its data rate to the furthest located users, so as to guarantee service quality to as many users as possible. Predictably, the more users in a multicast group, the lower data rate the base station can transmit. On the other hand, grouping more users together leads to a more efficient utilization of spectrum bandwidth, as these users are served simultaneously. This bring the interesting problem that presses for solution: how to group users in a cell into multicast groups and how to allocate a fixed amount of bandwidth resource to the groups, to achieve a good balance between throughput and fairness in multicast systems. In this paper, we formulate the united user grouping and bandwidth allocation strategy into a utility-based optimization problem. One method of signomial programming is used to solve the non-convex optimization problem. Numerical results will show that this suboptimal algorithm performs well even compared to the optimal one. Moreover, through theoretical analysis, we prove that the best user grouping and bandwidth allocation scheme of throughput maximization is to allocate the entire bandwidth to the unique group containing the users located within a ring-shaped region with an optimal outer radius r*.