A scalable WebRTC platform based on open technologies

International Conference on Computer, Information and Telecommunication Systems (7th, 2018, Colmar, Francia

Orchestrating Service Migration for Low Power MEC-Enabled IoT Devices

Multi-Access Edge Computing (MEC) is a key enabling technology for Fifth Generation (5G) mobile networks. MEC facilitates distributed cloud computing capabilities and information technology service environment for applications and services at the edges of mobile networks. This architectural modification serves to reduce congestion, latency, and improve the performance of such edge colocated applications and devices. In this paper, we demonstrate how reactive service migration can be orchestrated for low-power MEC-enabled Internet of Things (IoT) devices. Here, we use open-source Kubernetes as container orchestration system. Our demo is based on traditional client-server system from user equipment (UE) over Long Term Evolution (LTE) to the MEC server. As the use case scenario, we post-process live video received over web real-time communication (WebRTC). Next, we integrate orchestration by Kubernetes with S1 handovers, demonstrating MEC-based software defined network (SDN). Now, edge applications may reactively follow the UE within the radio access network (RAN), expediting low-latency. The collected data is used to analyze the benefits of the low-power MEC-enabled IoT device scheme, in which end-to-end (E2E) latency and power requirements of the UE are improved. We further discuss the challenges of implementing such schemes and future research directions therein

Communication System For Firefighters

Currently firefighters use two-way radios to communicate on the job, and they are forced to write reports based on their memory because there is not an easy way to record the communications between two-way radios. Firefighters need a system to automatically document what happened while they were responding to a call. To save them a significant amount of time when creating reports, our solution is to implement an application that allows firefighters to take pictures, record video and communicate in real time with their team of on-site responders. The proposed system will use a Wireless Local Area Network (WLAN) hosted on the fire truck itself to act as an access point (AP) to which the firefighters can connect. This AP will also save communication between firefighters to a local storage location. Upon return to the fire station, the AP will route all of the information stored locally to a larger database. For now, Wi-Fi will be our communication medium, with a prediction that our technology can eventually be extended to include radio signal

Website Technology Trends for Augmented Reality Development

Augmented reality (AR) is a technology that is gaining increasing attention from academics and industry. AR is relied upon to be an innovative technology to enrich ways of interacting with the physical and cyberspace around users that can enhance user experience in various fields. Platforms for AR applications are usually hardware based and mobile based, for mobile applications AR is usually based. AR-based hardware requires quite expensive support, this is seen from the rendering space requirements and this makes it inflexible while AR-based applications on mobile smartphones require large storage space and do not make it convenient for cross-platform use. Currently many researchers are trying to create and develop website-based AR, as a solution to the spread of AR to be flexible and save storage space, website technology development trends are used as a method for improving the performance of website-based AR. Other support comes from open-source software and more developer platforms and program courses for Web AR that are made public. This paper reviews the state-of-the-art, various methods, technologies and challenges of existing AR, this can be a trigger for more research interest and efforts to provide AR experienc

Opportunities and Challenges of Implementing P2P Streaming Applications in the Web

P2P applications are increasingly present on the web. We have identified a gap in current proposals when it comes to the use of traditional P2P overlays for real-time multimedia streaming. We analyze the possibilities and challenges to extend WebRTC in order to implement JavaScript APIs for P2P streaming algorithms

Design and implement a new mechanism for audio, video and screen recording based on WebRTC technology

Many years ago, Flash was essential in browsers to interact with the user media devices, such as a microphone and camera. Today, Web Real-Time Communication (WebRTC) technology has come to substitute the flash, so browsers do not need the flash to access media devices or establish their communication. However, WebRTC standards do not express precisely how browsers can record audios, videos or screen instead of describing getUserMedia API that enables a browser to access microphone and camera. The prime objective of this research is to create a new WebRTC recording mechanism to record audios, videos, and screen using Google Chrome, Firefox, and Opera. This experiment applied through Ethernet and Wireless of the Internet and 4G networks. Also, the recording mechanism of this research was obtained based on JavaScript Library for audio, video, screen (2D and 3D animation) recording. Besides, different audio and video codecs in Chrome, Firefox and Opera were utilised, such as VP8, VP9, and H264 for video, and Opus codec for audio. Not only but also, various bitrates (100 bytes bps, 1 Kbps, 100 Kbps, 1 MB bps, and 1 GB bps), different resolutions (1080p, 720p, 480p, and HD (3840* 2160)), and various frame-rates (fps) 5, 15, 24, 30 and 60 were considered and tested. Besides, an evaluation of recording mechanism, Quality of Experience (QoE) through actual users, resources, such as CPU performance was also done. In this paper, a novel implementation was accomplished over different networks, different browsers, various audio and video codecs, many peers, opening one or multi browsers at the same time, keep the streaming active as much as the user needs, save the record, using only audio and/or video recording as conferencing with full screen, etc