Performance Analysis of WebRTC and SIP for Video Conferencing
Navrattan Parmar1,  Virender Ranga2

1Navrattan Parmar,Department of Computer Engineering, National Institute of Technology, Kurukshetra, Haryana, India.

2Virender Ranga, Department of Computer Engineering, National Institute of Technology, Kurukshetra, Haryana, India.

Manuscript received on 05 August 2019 | Revised Manuscript received on 12 August 2019 | Manuscript Published on 26 August 2019 | PP: 679-686 | Volume-8 Issue-9S August 2019 | Retrieval Number: I101090789S19/19©BEIESP | DOI: 10.35940/ijitee.I1109.0789S19

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Abstract: 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.

Keywords: WebRTC, SIP, SDP, UDP, Codec, VoIP, Session Management, Internet Engineering Task Force (IETF), TLS, Channel Bitrate, Inter-Process Communication (IPC).
Scope of the Article: Industrial Internet of Things (IIoT)