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Reliability of Video Conferencing Applications Based on the WebRTC Standard Used on Different Web Platforms
Abstract
Video conferencing applications allow two or more people to connect and conduct video calls. During the call, people can see and hear each other. Therefore, to make a video call we need a device equipped with a camera and a microphone. One of the technologies that permits us to send video and audio stream is WebRTC (Web Real-Time Communication). In the paper, the applications using the WebRTC technology were tested for performance and reliability. Application performance refers to image quality, delay, number of frames depending on the speed of the Internet connection, and its type. The reliability was checked for establishing a connection in various environments (Web browser or mobile device) and maintaining the established connection during changing conditions (accidental interruption, interruption due to external circumstances or errors). The purpose of the experimental research was to examine the WebRTC technology in the context of using it to conduct video conferences on desktop and mobile platforms. The performance of this technology was tested on the native as well as developed implementations available by the most popular browsers.
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With the advancement in communication and development of technologies like VoIP and Video Conferencing, Web Real-Time Communication (WebRTC) is developed to communicate without plugins and stream the videos on a real time. It was initially developed by Web Consortium(W3C) and Internet Engineering Task Force (IETF). It allows to transfer videos and audios between different browsers. This research paper, analyse the parameters during the call in different browsers and conditions (number of end points). The concept of WebRTC is inspired from Session Initiation Protocol(SIP). It helps in the establishment of sessions and maintain it. It also supports data and message transmissions. It also works on remote location and different network transmission protocols. It also allows peer to peer communication. In this research work, we examine the behaviour of WebRTC and SIP during the call from different browsers. We examine the different parameters like packets sent, jitter, VO-Width and bandwidth during the call and call supported on cloud during our experimental work.
In the modern world and fast development of the internet, the connection among people is being very significant than ever, people are looking for new methods to do advance communication between them without any issue, real-time communication is one of this ways. WebRTC (Web real-time communication) is a futuristic technology that makes real-time communication capabilities in audio, video and data transmission possible in real time communication through web browsers by using JavaScript APIs (Application Programming Interface) without the plug-in. In this paper, we proposed a web peer to peer real time communication system that allows users to communicate with high-speed data transmission over the communication channel using WebRTC technology, HTML5 and use Node.js server address. The result shows that the system is stable, fully functional, safe and can use in a practical network to transmit and receive multimedia data in real time between users.
Enabling high Quality of Experience (QoE) with1video-conferencing solutions based on Web Real-Time Commu-2nication (WebRTC) protocols anywhere, anytime, is challenging3and triggers the exploration of new ways to gain QoE insights.4In this paper, we share initial observations from a within-5subjects experiment (N = 22) in which 2-party WebRTC-based6audiovisual conversations took place under varying technical7conditions. We collected self-report data, peripheral physiological8data and application-level performance statistics with the overall9aim of exploring their usefulness and compatibility for assessing10QoE in the context of WebRTC. Our preliminary findings11indicate that the varying quality is well-reflected in the self-12reported overall quality and annoyance. However, this does apply13when considering the emotional valence ratings and ECG data,14implying that other factors may play a role here.
In this paper, we investigate session-related per-formance statistics of a Web-based Real-Time Communication(WebRTC) application called appear.in. We explore the char-acteristics of these statistics and explore how they may relateto users’ Quality of Experience (QoE). More concretely, wehave run a series of tests involving two parties and accord-ing to different test scenarios, and collected real-time sessionstatistics by means of Google Chrome’s WebRTC-internals tool.Despite the fact that the Chrome statistics have a number oflimitations, our observations indicate that they are useful forQoE research when these limitations are known and carefullyhandled when performing post-processing analysis. The resultsfrom our initial tests show that a combination of performanceindicators measured at the sender’s and receiver’s end may helpto identify severe video freezes (being an important QoE killer)in the context of WebRTC-based video communication. In thispaper the performance indicators used are significant drops indata rate, non-zero packet loss ratios, non-zero PLI values, andnon-zero bucket delay.
Video QoE killer and performance statistics in WebRTC-based video communication. Available from: https://www.researchgate.net/publication/307940053_Video_QoE_killer_and_performance_statistics_in_WebRTC-based_video_communication [accessed Feb 12 2018].
The continuous advancement in communication technology has introduced many video communication applications. The rise in demand for these applications has brought about an increase in bandwidth consumption. Despite the ability of the new generation of technology to render this need, challenges on transmitting a video with good quality over networks still exists. One of these is the variation in delay caused by congestion in the links of the network. In real-time transmission, this delay can cause a packet to be dropped, leading to an error in the video stream. This error can propagate to the subsequent frames, further degrading the quality of video at the receiver end. One major cause of error propagation is the loss of packets due to an extensive delay. In this paper, a multi-hop wired network was simulated to demonstrate packet loss due to packet drop caused by queuing on the hops; and the effects of packet drop in video transmission on the quality of video were observed. A maximum threshold end-to-end delay had been observed wherein the packet is dropped when this value of delay is exceeded. The dropped packet had been correlated to the number of queued packets on the link where it was dropped. With the simulated packet loss pattern used on video transmission, the quality of video between lossless video transmission and that with packet loss and the effects of packet loss between different bit rate and motion complexity had been observed and differentiated.
We expect that WebRTC will experience high adoption rate as a peer-to-peer realtime communication standard for browsers. WebRTC allows direct media and data transport between browsers without having to go through a web server. In this study, we use a black-box testing technique to evaluate, via PESQ, the voice quality of WebRTC sessions under varying network delay and jitter. Network emulators are employed to implement the delay and jitter variations. Our results highlight the dangers of black-box testing, whereby test-bed issues can result in very misleading results. This is especially the case when executed on a single machine. This paper also provides an extendable baseline methodology for WebRTC centric research.
Revealing the dark side of WebRTC statistics collected by Google Chrome
- D Ammar
- K De Moor
- P E Heegaard
- M Fiedler
- M Xie