Open-source RTP Library for End-to-End Encrypted Real-Time Video Streaming Applications

Information security has become a key success factor for streaming media applications that are increasingly vulnerable to wiretapping, message forgery, data tampering, hacking, and other possible cyberattacks. This paper addresses the existing security risks in real-time video streaming by introducing a new security extension to our uvgRTP open-source Real-time Transport Protocol (RTP) library. The proposed solution improves content integrity and privacy by adopting Secure RTP (SRTP) and Zimmermann RTP (ZRTP) for media End-to-End Encryption (E2EE). These new security mechanisms make uvgRTP the first open-source library that supports on-the-fly encrypted AVC, HEVC, and VVC video streaming. Our performance results on Intel Core i7-4770 processor show that uvgRTP is able to transport encrypted 8K VVC video at up to 187 fps and 8K HEVC video at up to 120 fps over a 10 Gbps Local Area Network (LAN). The achieved transfer rate for encrypted HEVC video is 50% higher and latency 86% lower than the respective performance values of FFmpeg in unencrypted HEVC streaming. These top streaming speed results with state-of-the-art video codec support, advanced encryption mechanisms, and the permissive BSD license make uvgRTP an attractive solution for a broad range of commercial and academic streaming media applications.acceptedVersionPeer reviewe

uvgRTP 2.0: Open-Source RTP Library For Real-Time VVC/HEVC Streaming

Real-time video transport plays a central role in various interactive and streaming media applications. This paper presents a new release of our open-source Real-time Transport Protocol (RTP) library called uvgRTP (github.com/ultravideo/uvgRTP) that is designed for economic video and audio transmission in real time. It is the first public library that comes with built-in support for modern VVC, HEVC, and AVC video codecs and Opus audio codec. It can also be tailored to diversified media formats with an easy-to-use generic API. According to our experiments, uvgRTP can stream 8K VVC video at 300 fps with an average round-trip latency of 4.9 ms over a 10 Gbit link. This cross-platform library can be run on Windows and Linux operating systems and the permissive BSD 2-Clause license makes it accessible to a broad range of commercial and academic streaming media applications.acceptedVersionPeer reviewe

A Prototype Modelling of Ebers for Video Transmission in Wireless Adhoc Network

Provisioning of video streaming over ad hoc wireless networks exhibits challenges associated with high packet loss rates and are delay sensitive Excessive packet loss can cause significant degradation in quality of video perceived by users of real-time video applications The recent studies suggest that Forward Error Correction FEC is a good technique for decreasing the negative impact of packet loss on video quality in error control scheme This paper introduces an Estimation based Error Reduction Scheme EBERS to support video communication in ad hoc wireless networks The EBERS considers a frame estimation parameter to support varied bandwidths and attain the delay requirements to support video communication It is also responsible for improvising the QoS offered The EBERS considers layered and embodies distortion limiting features owing to which reduced forward error correction is achieved thus obtaining reduced frame errors transmission errors and retransmission of frames Thereby obtaining high degree of quality of service QoS The comparative study conducted proves the efficiency of the EBERS scheme over the existing mechanism

Video QoS/QoE over IEEE802.11n/ac: A Contemporary Survey

The demand for video applications over wireless networks has tremendously increased, and IEEE 802.11 standards have provided higher support for video transmission. However, providing Quality of Service (QoS) and Quality of Experience (QoE) for video over WLAN is still a challenge due to the error sensitivity of compressed video and dynamic channels. This thesis presents a contemporary survey study on video QoS/QoE over WLAN issues and solutions. The objective of the study is to provide an overview of the issues by conducting a background study on the video codecs and their features and characteristics, followed by studying QoS and QoE support in IEEE 802.11 standards. Since IEEE 802.11n is the current standard that is mostly deployed worldwide and IEEE 802.11ac is the upcoming standard, this survey study aims to investigate the most recent video QoS/QoE solutions based on these two standards. The solutions are divided into two broad categories, academic solutions, and vendor solutions. Academic solutions are mostly based on three main layers, namely Application, Media Access Control (MAC) and Physical (PHY) which are further divided into two major categories, single-layer solutions, and cross-layer solutions. Single-layer solutions are those which focus on a single layer to enhance the video transmission performance over WLAN. Cross-layer solutions involve two or more layers to provide a single QoS solution for video over WLAN. This thesis has also presented and technically analyzed QoS solutions by three popular vendors. This thesis concludes that single-layer solutions are not directly related to video QoS/QoE, and cross-layer solutions are performing better than single-layer solutions, but they are much more complicated and not easy to be implemented. Most vendors rely on their network infrastructure to provide QoS for multimedia applications. They have their techniques and mechanisms, but the concept of providing QoS/QoE for video is almost the same because they are using the same standards and rely on Wi-Fi Multimedia (WMM) to provide QoS

Implementation of 4kUHD HEVC-content transmission

The Internet of things (IoT) has received a great deal of attention in recent years, and is still being approached with a wide range of views. At the same time, video data now accounts for over half of the internet traffic. With the current availability of beyond high definition, it is worth understanding the performance effects, especially for real-time applications. High Efficiency Video Coding (HEVC) aims to provide reduction in bandwidth utilisation while maintaining perceived video quality in comparison with its predecessor codecs. Its adoption aims to provide for areas such as television broadcast, multimedia streaming/storage, and mobile communications with significant improvements. Although there have been attempts at HEVC streaming, the literature/implementations offered do not take into consideration changes in the HEVC specifications. Beyond this point, it seems little research exists on real-time HEVC coded content live streaming. Our contribution fills this current gap in enabling compliant and real-time networked HEVC visual applications. This is done implementing a technique for real-time HEVC encapsulation in MPEG-2 Transmission Stream (MPEG-2 TS) and HTTP Live Streaming (HLS), thereby removing the need for multi-platform clients to receive and decode HEVC streams. It is taken further by evaluating the transmission of 4k UHDTV HEVC-coded content in a typical wireless environment using both computers and mobile devices, while considering well-known factors such as obstruction, interference and other unseen factors that affect the network performance and video quality. Our results suggest that 4kUHD can be streamed at 13.5 Mb/s, and can be delivered to multiple devices without loss in perceived quality

A Simulation Tool for Evaluating Video Streaming Architectures in Vehicular Network Scenarios

An integrated simulation tool called Video Delivery Simulation Framework over Vehicular Networks (VDSF-VN) is presented. This framework is intended to allow users to conduct experiments related to video transmission in vehicular networks by means of simulation. Research on this topic requires the use of many independent tools, such as traffic and network simulators, intermediate frameworks, video encoders and decoders, converters, platform-dependent scripting languages, data visualisation packages and spreadsheets, and some other tasks are performed manually. The lack of tools necessary to carry out all these tasks in an integrated and efficient way formed the motivation for the development of the VDSF-VN framework. It is managed via two user-friendly applications, GatcomSUMO and GatcomVideo, which allow all the necessary tasks to be accomplished. The first is primarily used to build the network scenario and set up the traffic flows, whereas the second involves the delivery process of the whole video, encoding/decoding video, running simulations, and processing all the experimental results to automatically provide the requested figures, tables and reports. This multiplatform framework is intended to fill the existing gap in this field, and has been successfully used in several experimental tests of vehicular network

Understanding Timelines within MPEG Standards

(c) 2016 IEEE. Personal use of this material is permitted. Permission from IEEE must be obtained for all other uses, in any current or future media, including reprinting/republishing this material for advertising 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.Nowadays, media content can be delivered via diverse broadband and broadcast technologies. Although these different technologies have somehow become rivals, their coordinated usage and convergence, by leveraging of their strengths and complementary characteristics, can bring many benefits to both operators and customers. For example, broadcast TV content can be augmented by on-demand broadband media content to provide enriched and personalized services, such as multi-view TV, audio language selection, and inclusion of real-time web feeds. A piece of evidence is the recent Hybrid Broadcast Broadband TV (HbbTV) standard, which aims at harmonizing the delivery and consumption of (hybrid) broadcast and broadband TV content. A key challenge in these emerging scenarios is the synchronization between the involved media streams, which can be originated by the same or different sources, and delivered via the same or different technologies. To enable synchronized (hybrid) media delivery services, some mechanisms providing timelines at the source side are necessary to accurately time align the involved media streams at the receiver-side. This paper provides a comprehensive review of how clock references (timing) and timestamps (time) are conveyed and interpreted when using the most widespread delivery technologies, such as DVB, RTP/RTCP and MPEG standards (e.g., MPEG-2, MPEG-4, MPEG-DASH, and MMT). It is particularly focused on the format, resolution, frequency, and the position within the bitstream of the fields conveying timing information, as well as on the involved components and packetization aspects. Finally, it provides a survey of proofs of concepts making use of these synchronization related mechanisms. This complete and thorough source of information can be very useful for scholars and practitioners interested in media services with synchronization demands.This work has been funded, partially, by the "Fondo Europeo de Desarrollo Regional" (FEDER) and the Spanish Ministry of Economy and Competitiveness, under its R&D&i Support Program in project with ref TEC2013-45492-R.Yuste, LB.; Boronat Segui, F.; Montagut Climent, MA.; Melvin, H. (2015). Understanding Timelines within MPEG Standards. Communications Surveys and Tutorials, IEEE Communications Society. 18(1):368-400. https://doi.org/10.1109/COMST.2015.2488483S36840018