September 2024 Summaries

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.

The text discusses a comparison of call quality between two video conferencing platforms, 100ms and Provider Z. An independent lab conducted tests under various network conditions to evaluate the performance of both platforms. Results show that while video quality is comparable, 100ms outperforms Provider Z in terms of audio quality, frames per second, and overall latency. In poor network conditions, 100ms experiences less freezing and lower audio delay compared to Provider Z. The study concludes that 100ms provides a more reliable live video platform even when the network is unreliable.