Measuring WebRTC Call Quality - Part 1

This blog discusses the importance of measuring WebRTC call quality and highlights key metrics provided by the getStats() API that help diagnose and improve call quality issues. The sender-side metrics explored include frame width, frame height, frames per second, bytesSent, outgoing bitrate, packetsLost, round trip time (RTT), jitter, totalPacketSendDelay, availableOutgoingBitrate, and qualityLimitationDurations. These metrics are crucial for understanding the factors affecting call quality and optimizing network resources to ensure smooth audio/video playback and a consistent user experience.