<![CDATA[Video and audio communications have become part of all areas of work. Two real-timecommunication protocols commonly used for IP-based video and audio communicationsare Session Initiation Protocol (SIP) and real-time web communications (WebRTC). Bothprotocols have been widely used in softphone and video conferencing applications. Themain objective of this research is to make an analysis of the performance of a client serverapplication for video and audio communications developed by SIP and WebRTC. The SIPsystem consists of a softphone on the client side using Bria and a FreePBX server, forWebRTCapplications, using JavaScript and a server at Node.js. The results showed that the WebRTCaudio and video communication provided better quality in terms of PSNR. This is due tothe different codecs used between WebRTC and SIP. WebRTC uses VP8 as video codec, SIPuses H.246 as video codec, WebRTC uses G.711 as audio codec, and implemented SIP usesG.729 as audio codec.]]>
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