Open AccessPerformance Analysis of WebRTC and SIP for Video Conferencing
Publication year - 2019
Publication title -
international journal of innovative technology and exploring engineering
Language(s) - English
Resource type - Journals
ISSN - 2278-3075
DOI - 10.35940/ijitee.i1109.0789s19
Subject(s) - webrtc , computer science , voice over ip , session initiation protocol , jitter , sip trunking , videoconferencing , the internet , computer network , network packet , plug in , session (web analytics) , multimedia , cloud computing , world wide web , server , operating system , telecommunications
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.