| Resumo: | This dissertation addresses the challenge of developing a video call system capable of supporting both Android mobile devices and fixed computers. Addi tionally, it analyses the quality of video achieved and its variation in the presence of network bandwidth and packet loss constraints. A prototype of a video call system was implemented using a web application and the Web Real-Time Communication (WebRTC) library. Clients use WebRTC to stream video over a Traversal Using Relays around NAT (TURN) relay server, allowing them to send video to any terminal connected to the Internet. Signalling was implemented using WebSockets and a Node.js server. A quality testing prototype was also implemented, which supports sending pre-recorded videos and capturing and storing video recordings at the sender and receiver. The Video Multimethod Assessment Fusion (VMAF) metric was used as the main video quality metric, based on the comparison between the transmitted and received videos. The quality of a video encoded using the open source video encoder VP8 was analysed in constrained network setups. The results measured the video quality degradation and percentage of received frames, showing that the system is resilient to some bandwidth strangulation and packet loss, although with a noticeable video quality degradation.
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