Study of the Effect of Security on Quality of Service on a WebRTC Framework for Videocalls

Web Real-Time Communication (WebRTC) is a promising new standard and technology stack, providing full audio/video communications in a secured solution. Organizations implementing such technology deal with both quality of service and security demands, therefore it is mandatory to investigate the impact of QoS parameters when applying security mechanisms in multimedia services for the case of videocalls. This work presents a study of quality of service indicators such as jitter, delay between RTP (Real Time Protocol) packets, establishment and release time for three levels of security, considering the effect over signaling and media planes of the videocall. Four scenarios were implemented in two groups: the first one consisted of a LAN with a Laptop and a PC with WebRTC clients or between a Smartphone and a PC as well. The second one consisted of a DSL WAN with a Laptop and a PC with WebRTC clients or between a Smartphone and a PC as well. The three levels of security for each scenario were implemented as follows: the first one without security, the second one with TLS in signaling, and the third with both TLS in signaling and DTLS in media traffic. Codecs employed were VP8 for video and OPUS for audio on every testbed over WebRTC with EasyRTC framework. The results show, in general terms that QoS media indicators were on the recommended levels according to ITU and IETF. Establishing and liberation time were degraded. In some cases there was an improvement, as in the case of jitter due to the use of RTP (Real Time Transport) protocol for audio. We recommend the use of machine learning algorithm K-means for detecting clusters between different QoS indicator for the three levels of security just to detect with more accuracy the impact for each level.