Enhancement of Adaptive Forward Error Correction Mechanism for Video Transmission Over Wireless Local Area Network

Video transmission over the wireless network faces many challenges. The most critical challenge is related to packet loss. To overcome the problem of packet loss, Forward Error Correction is used by adding extra packets known as redundant packet or parity packet. Currently, FEC mechanisms have been adopted together with Automatic Repeat reQuest (ARQ) mechanism to overcome packet losses and avoid network congestion in various wireless network conditions. The number of FEC packets need to be generated effectively because wireless network usually has varying network conditions. In the current Adaptive FEC mechanism, the FEC packets are decided by the average queue length and average packet retransmission times. The Adaptive FEC mechanisms have been proposed to suit the network condition by generating FEC packets adaptively in the wireless network. However, the current Adaptive FEC mechanism has some major drawbacks such as the reduction of recovery performance which injects too many excessive FEC packets into the network. This is not flexible enough to adapt with varying wireless network condition. Therefore, the enhancement of Adaptive FEC mechanism (AFEC) known as Enhanced Adaptive FEC (EnAFEC) has been proposed. The aim is to improve recovery performance on the current Adaptive FEC mechanism by injecting FEC packets dynamically based on varying wireless network conditions. The EnAFEC mechanism is implemented in the simulation environment using Network Simulator 2 (NS-2). Performance evaluations are also carried out. The EnAFEC was tested with the random uniform error model. The results from experiments and performance analyses showed that EnAFEC mechanism outperformed the other Adaptive FEC mechanism in terms of recovery efficiency. Based on the findings, the optimal amount of FEC generated by EnAFEC mechanism can recover high packet loss and produce good video quality

Reducing Internet Latency : A Survey of Techniques and their Merit

Bob Briscoe, Anna Brunstrom, Andreas Petlund, David Hayes, David Ros, Ing-Jyh Tsang, Stein Gjessing, Gorry Fairhurst, Carsten Griwodz, Michael WelzlPeer reviewedPreprin

Towards video streaming in IoT environments: vehicular communication perspective

Multimedia oriented Internet of Things (IoT) enables pervasive and real-time communication of video, audio and image data among devices in an immediate surroundings. Today's vehicles have the capability of supporting real time multimedia acquisition. Vehicles with high illuminating infrared cameras and customized sensors can communicate with other on-road devices using dedicated short-range communication (DSRC) and 5G enabled communication technologies. Real time incidence of both urban and highway vehicular traffic environment can be captured and transmitted using vehicle-to-vehicle and vehicle-to-infrastructure communication modes. Video streaming in vehicular IoT (VSV-IoT) environments is in growing stage with several challenges that need to be addressed ranging from limited resources in IoT devices, intermittent connection in vehicular networks, heterogeneous devices, dynamism and scalability in video encoding, bandwidth underutilization in video delivery, and attaining application-precise quality of service in video streaming. In this context, this paper presents a comprehensive review on video streaming in IoT environments focusing on vehicular communication perspective. Specifically, significance of video streaming in vehicular IoT environments is highlighted focusing on integration of vehicular communication with 5G enabled IoT technologies, and smart city oriented application areas for VSV-IoT. A taxonomy is presented for the classification of related literature on video streaming in vehicular network environments. Following the taxonomy, critical review of literature is performed focusing on major functional model, strengths and weaknesses. Metrics for video streaming in vehicular IoT environments are derived and comparatively analyzed in terms of their usage and evaluation capabilities. Open research challenges in VSV-IoT are identified as future directions of research in the area. The survey would benefit both IoT and vehicle industry practitioners and researchers, in terms of augmenting understanding of vehicular video streaming and its IoT related trends and issues

CLOUD LIVE VIDEO TRANSFER

As multimedia content continues to grow, considerations for more effective storage options, like cloud technologies, become apparent. While video has become a mainstream media source on the web, live video streaming is growing as a prominent player in the modern marketplace for both businesses and individuals. For instance, a business owner may want to oversee operations while he or she is away, or an individual may want to surveillance their property. In this work, we propose Cloud Live Video Streaming (CLVS) - a very efficient method to stream live video that creates a separate pricing model from modern video streaming services. The key component to CLVS is Amazon Simple Storage Service (S3), which is used to store video segments and metadata. By using S3, CLVS employs what is referred to as a ”serverless” design by removing the need to stream video through an intermediary server. CLVS also removes the need for third party accounts and license agreements. We implement a prototype of CLVS and compare it with an existing commercial video streaming software - Wowza Streaming Engine. As live video streaming becomes more common, alternative and cost effective solutions are essential

Control Mechanisms and Recovery Techniques for Real-Time Data Transmission Over the Internet.

Streaming multimedia content with UDP has become popular over distributed systems such as an Internet. This may encounter many losses due to dropped packets or late arrivals at destination since UDP can only provide best effort delivery. Even UDP doesn't have any self-recovery mechanism from congestion collapse or bursty loss to inform sender of the data to adjust future transmission rate of data like in TCP. So there is a need to incorporate various control schemes like forward error control, interleaving, and congestion control and error concealment into real-time transmission to prevent from effect of losses. Loss can be repaired by retransmission if roundtrip delay is allowed, otherwise error concealment techniques will be used based on the type and amount of loss. This paper implements the interleaving technique with packet spacing of varying interleaver block size for protecting real-time data from loss and its effect during transformation across the Internet. The packets are interleaved and maintain some time gap between two consecutive packets before being transmitted into the Internet. Thus loss of packets can be reduced from congestion and preventing loss of consecutive packets of information when a burst of several packets are lost. Several experiments have been conducted with video data for analysis of proposed model

Error and Congestion Resilient Video Streaming over Broadband Wireless

In this paper, error resilience is achieved by adaptive, application-layer rateless channel coding, which is used to protect H.264/Advanced Video Coding (AVC) codec data-partitioned videos. A packetization strategy is an effective tool to control error rates and, in the paper, source-coded data partitioning serves to allocate smaller packets to more important compressed video data. The scheme for doing this is applied to real-time streaming across a broadband wireless link. The advantages of rateless code rate adaptivity are then demonstrated in the paper. Because the data partitions of a video slice are each assigned to different network packets, in congestion-prone wireless networks the increased number of packets per slice and their size disparity may increase the packet loss rate from buffer overflows. As a form of congestion resilience, this paper recommends packet-size dependent scheduling as a relatively simple way of alleviating the buffer-overflow problem arising from data-partitioned packets. The paper also contributes an analysis of data partitioning and packet sizes as a prelude to considering scheduling regimes. The combination of adaptive channel coding and prioritized packetization for error resilience with packet-size dependent packet scheduling results in a robust streaming scheme specialized for broadband wireless and real-time streaming applications such as video conferencing, video telephony, and telemedicine

Squash: low latency multi-path video streaming using multi-bitrate encoding

The demand for low latency video streaming has dramatically increased as live video streaming applications, such as Twitch and Youtube Live, are becoming more popular. According to the 2021 Bitmovin video developer report, the biggest challenge that video developers are experiencing today is providing low latency video streaming. One of the most common on-site live streaming methods is using a wireless LTE network. There have been many approaches for characterizing wireless links and accurately measuring available bandwidth to provide low latency streaming over a wireless LTE network link. However, even with fine-grained bandwidth estimation, video streaming on a single LTE link is still susceptible to unexpected network delay from a sudden drop in available bandwidth or temporal disconnection. People can utilize multiple wireless LTE links to overcome the limitations of using a single LTE link for low latency video streaming. Using multiple links can enhance video quality through increased bandwidth and resilience. However, multi-homed low latency video streaming protocols may achieve lower video quality than single-homed protocols when a frame is split and sent over more than one link. Suppose one of the links becomes congested or gets disconnected. In that case, the part of the frame sent on stable links must wait until the packets sent on the problematic link are re-transmitted through another link. Re-transmission requires at least one extra round trip time. A video player may skip the late frame or serve only the received part of the frame due to the re-transmission delay. Ferlin et al. suggest using Forward Error Correction (FEC) on Multipath TCP (MPTCP) to reduce re-transmission delay. However, FEC is not helpful in the event of a significant bandwidth drop. If the sender does not use sufficient redundancy to handle a significant bandwidth drop, the receiver will not receive enough blocks to decode the video data. FEC requires using a large portion of the network bandwidth for redundancy to handle significant bandwidth drops even when the links are stable. In this thesis, I present Squash, a low latency video transport protocol that encodes each frame at multiple bitrates and sends them across different links to minimize video stream disruption in the event of unexpected bandwidth drops. The encoder encodes a frame into multiple different bitrates, which are high-bitrate and low-bitrate. When a high- bitrate frame cannot arrive on time due to congestion from an unexpected drop in available bandwidth, the low-bitrate frame is used to replace the missing frame. This is because the low-bitrate frame is smaller and is sent on the links that are disjoint from those used by the high-bitrate frame. To the best of my knowledge, Squash is the first architecture that uses multi-bitrate frames to increase resilience against unexpected bandwidth drops in low latency video streaming over multiple wireless LTE links. In emulated wireless LTE network environment using Mahimahi network traces, the average SSIM of the video streamed on Squash is 13 – 58% higher than that streamed on the baseline protocol, which is designed in the same manner as Squash except that it employs single-frame encoding

Enabling E2E reliable communications with adaptive re-encoding over delay tolerant networks

Reliable end-to-end (E2E) communication in Delay Tolerant Network (DTN) is a challenging task due to long delay and frequent link disruptions. To enable reliability, the IETF is currently looking at strategies to integrate erasure coding mechanisms inside DTN architecture. The objective is to extend the ability of the existing DTN bundle fragmentation mechanism to support cases where bundles have a high probability of being lost. To date, discussions agree that an intermediate node can re-encode bundles, leaving all decoding process at the destination node in order to let intermediate node operations be as simple as possible. We propose to study and analyze possible re-encoding strategies at intermediate nodes using an on-the-fly coding paradigm. We also investigate how re-encoding and acknowledgment strategies based on this coding scheme would enable reliable E2E communication. Finally, we propose an adaptive mechanism with low complexity that deals with both re-routing events and network dynamics which are common in the context of DTN. Simulation results show that re-encoding at the relay and the adaptive mechanism allows a significant reduction in terms of network overhead injected by erasure codes while ensuring the E2E reliability