The Quality Comparison of WebRTC and SIP Audio and Video Communications with PSNR
Video and audio communications have become part of all areas of work. Two real-time
communication protocols commonly used for IP-based video and audio communications
are Session Initiation Protocol (SIP) and real-time web communications (WebRTC). Both
protocols have been widely used in softphone and video conferencing applications. The
main objective of this research is to make an analysis of the performance of a client server
application for video and audio communications developed by SIP and WebRTC. The SIP
system consists of a softphone on the client side using Bria and a FreePBX server, forWebRTC
audio and video communication provided better quality in terms of PSNR. This is due to
the different codecs used between WebRTC and SIP. WebRTC uses VP8 as video codec, SIP
uses H.246 as video codec, WebRTC uses G.711 as audio codec, and implemented SIP uses
G.729 as audio codec.
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