Multimedia delivery in the future internet

The term “Networked Media” implies that all kinds of media including text, image, 3D graphics, audio and video are produced, distributed, shared, managed and consumed on-line through various networks, like the Internet, Fiber, WiFi, WiMAX, GPRS, 3G and so on, in a convergent manner [1]. This white paper is the contribution of the Media Delivery Platform (MDP) cluster and aims to cover the Networked challenges of the Networked Media in the transition to the Future of the Internet. Internet has evolved and changed the way we work and live. End users of the Internet have been confronted with a bewildering range of media, services and applications and of technological innovations concerning media formats, wireless networks, terminal types and capabilities. And there is little evidence that the pace of this innovation is slowing. Today, over one billion of users access the Internet on regular basis, more than 100 million users have downloaded at least one (multi)media file and over 47 millions of them do so regularly, searching in more than 160 Exabytes1 of content. In the near future these numbers are expected to exponentially rise. It is expected that the Internet content will be increased by at least a factor of 6, rising to more than 990 Exabytes before 2012, fuelled mainly by the users themselves. Moreover, it is envisaged that in a near- to mid-term future, the Internet will provide the means to share and distribute (new) multimedia content and services with superior quality and striking flexibility, in a trusted and personalized way, improving citizens’ quality of life, working conditions, edutainment and safety. In this evolving environment, new transport protocols, new multimedia encoding schemes, cross-layer inthe network adaptation, machine-to-machine communication (including RFIDs), rich 3D content as well as community networks and the use of peer-to-peer (P2P) overlays are expected to generate new models of interaction and cooperation, and be able to support enhanced perceived quality-of-experience (PQoE) and innovative applications “on the move”, like virtual collaboration environments, personalised services/ media, virtual sport groups, on-line gaming, edutainment. In this context, the interaction with content combined with interactive/multimedia search capabilities across distributed repositories, opportunistic P2P networks and the dynamic adaptation to the characteristics of diverse mobile terminals are expected to contribute towards such a vision. Based on work that has taken place in a number of EC co-funded projects, in Framework Program 6 (FP6) and Framework Program 7 (FP7), a group of experts and technology visionaries have voluntarily contributed in this white paper aiming to describe the status, the state-of-the art, the challenges and the way ahead in the area of Content Aware media delivery platforms

A Review of Error Resilience Techniques in Video Streaming

Abstract-Delivering video data of satisfactory quality over unreliable networks -such as the internet or wireless networks -is a demanding area which has received significant attention of the research community over the past few years. Given the fact that packet loss is inevitable and therefore the presence of errors granted, the effort is directed towards limiting the effect of these errors. A number of techniques have been developed to address this issue. This paper aims to summarize the most significant approaches for: error resilience, error concealment and joint encoder-decoder error control techniques, and to provide a thorough discussion of the benefits and drawbacks of these error control methods. Furthermore, two case studies of error resilience utilization are presented, namely Ad-hoc networks and Multimedia Broadcast Multiple Services (MBMS)

Structured Random Linear Codes (SRLC): Bridging the Gap between Block and Convolutional Codes

Several types of AL-FEC (Application-Level FEC) codes for the Packet Erasure Channel exist. Random Linear Codes (RLC), where redundancy packets consist of random linear combinations of source packets over a certain finite field, are a simple yet efficient coding technique, for instance massively used for Network Coding applications. However the price to pay is a high encoding and decoding complexity, especially when working on $GF(2^8)$, which seriously limits the number of packets in the encoding window. On the opposite, structured block codes have been designed for situations where the set of source packets is known in advance, for instance with file transfer applications. Here the encoding and decoding complexity is controlled, even for huge block sizes, thanks to the sparse nature of the code and advanced decoding techniques that exploit this sparseness (e.g., Structured Gaussian Elimination). But their design also prevents their use in convolutional use-cases featuring an encoding window that slides over a continuous set of incoming packets. In this work we try to bridge the gap between these two code classes, bringing some structure to RLC codes in order to enlarge the use-cases where they can be efficiently used: in convolutional mode (as any RLC code), but also in block mode with either tiny, medium or large block sizes. We also demonstrate how to design compact signaling for these codes (for encoder/decoder synchronization), which is an essential practical aspect.Comment: 7 pages, 12 figure

Overview of evolved Multimedia Broadcast Multicast Services (eMBMS)

MBMS was introduced as a service to optimize the dissemination of common interest multimedia content. Recently, it evolved to eMBMS based on LTE-centered flexibilities. However, launch of eMBMS over LTE may support new services e.g. pushed content for M2M services and delivery of premium content to the users enjoying secured QoS. This document primarily focusses on the rules, procedures and architecture supporting MBMS based data exchanges, which have not seen any major changes since Release 9

5G New Radio for Terrestrial Broadcast: A Forward-Looking Approach for NR-MBMS

"© 2019 IEEE. Personal use of this material is permitted. Permissíon from IEEE must be obtained for all other uses, in any current or future media, including reprinting/republishing this material for advertisíng or promotional purposes, creating new collective works, for resale or redistribution to servers or lists, or reuse of any copyrighted component of this work in other works."[EN] 3GPP LTE eMBMS release (Rel-) 14, also referred to as further evolved multimedia broadcast multicast service (FeMBMS) or enhanced TV (EnTV), is the first mobile broadband technology standard to incorporate a transmission mode designed to deliver terrestrial broadcast services from conventional high power high tower (HPHT) broadcast infrastructure. With respect to the physical layer, the main improvements in FeMBMS are the support of larger inter-site distance for single frequency networks (SFNs) and the ability to allocate 100% of a carrier's resources to the broadcast payload, with self-contained signaling in the downlink. From the system architecture perspective, a receive-only mode enables free-to-air (FTA) reception with no need for an uplink or SIM card, thus receiving content without user equipment registration with a network. These functionalities are only available in the LTE advanced pro specifications as 5G new radio (NR), standardized in 3GPP from Rel-15, has so far focused entirely on unicast. This paper outlines a physical layer design for NR-MBMS, a system derived, with minor modifications, from the 5G-NR specifications, and suitable for the transmission of linear TV and radio services in either single-cell or SFN operation. This paper evaluates the NR-MBMS proposition and compares it to LTE-based FeMBMS in terms of flexibility, performance, capacity, and coverage.This work was supported in part by the European Commission through the 5G-PPP Project 5G-Xcast (H2020-ICT-2016-2 call) under Grant 761498.Gimenez, JJ.; Carcel, JL.; Fuentes, M.; Garro, E.; Elliott, S.; Vargas, D.; Menzel, C.... (2019). 5G New Radio for Terrestrial Broadcast: A Forward-Looking Approach for NR-MBMS. IEEE Transactions on Broadcasting. 65(2):356-368. https://doi.org/10.1109/TBC.2019.291211735636865

Reliable and Low-Latency Fronthaul for Tactile Internet Applications

With the emergence of Cloud-RAN as one of the dominant architectural solutions for next-generation mobile networks, the reliability and latency on the fronthaul (FH) segment become critical performance metrics for applications such as the Tactile Internet. Ensuring FH performance is further complicated by the switch from point-to-point dedicated FH links to packet-based multi-hop FH networks. This change is largely justified by the fact that packet-based fronthauling allows the deployment of FH networks on the existing Ethernet infrastructure. This paper proposes to improve reliability and latency of packet-based fronthauling by means of multi-path diversity and erasure coding of the MAC frames transported by the FH network. Under a probabilistic model that assumes a single service, the average latency required to obtain reliable FH transport and the reliability-latency trade-off are first investigated. The analytical results are then validated and complemented by a numerical study that accounts for the coexistence of enhanced Mobile BroadBand (eMBB) and Ultra-Reliable Low-Latency (URLLC) services in 5G networks by comparing orthogonal and non-orthogonal sharing of FH resources.Comment: 11pages, 13 figures, 3 bio photo

Optimum Physical-Layer Frame Size for Maximising the Application-Layer Rateless Code’s Effective Throughput

The tolerable packet-loss ratio of an Internet Protocol (IP) based wireless networks varies according to the specific services considered. File transfer for example must be error free but tolerates higher delays, whereas maintaining a low delay is typically more important in interactive Voice Over IP (VOIP) or video services. Classic Forward Error Correction (FEC) may be applied to the data to provide resilience against bit errors. A wireless IP network provides the opportunity for the inclusion of FEC at the physical, transport and application layers. The demarcation between the analogue and digital domain imposed at the Physical layer (PHY) predetermines the nature of the FEC scheme implemented at the various layers. At the PHY individual packets may be offered FEC protection, which increases the likelihood of their error-free insertion into the protocol stack. Higher layers receive packets that are error free and the purpose of a FEC scheme implemented here is to regenerate any missing packets obliterated for example by the Binary Erasure Channel (BEC) of the IP network’s routers. A rateless code may be beneficially employed at a higher Open Systems Interconnection (OSI) layer for replenishing the obliterated packets, but unless the characteristics of the channel are considered, the ultimate rate achieved by such a code may be compromised, as shown in this contribution

Adaptive unicast video streaming with rateless codes and feedback.

Video streaming over the Internet and packet-based wireless networks is sensitive to packet loss, which can severely damage the quality of the received video. To protect the transmitted video data against packet loss, application-layer forward error correction (FEC) is commonly used. Typically, for a given source block, the channel code rate is fixed in advance according to an estimation of the packet loss rate. However, since network conditions are difficult to predict, determining the right amount of redundancy introduced by the channel encoder is not obvious. To address this problem, we consider a general framework where the sender applies rateless erasure coding to every source block and 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 design transmission strategies that aim at minimizing the expected bandwidth usage while ensuring successful decoding subject to an upper bound on the packet loss rate. In real simulations over the Internet, our solution outperformed standard FEC and hybrid ARQ approaches. For the QCIF Foreman sequence compressed with the H.264 video coder, the gain in average peak signal to noise ratio over the best previous scheme exceeded 3.5 decibels at 90 kilobits per second.DFG (German Research Foundation