Basics of WebRTC getStats() API
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The document discusses the WebRTC getStats API and its purpose in monitoring media quality through various metrics such as packet loss, jitter, and media rendering statistics. It highlights potential issues like lost or belated packets leading to audio and video disruptions. The statistics API is essential for understanding connection disruptions and network congestion in real-time communication scenarios.
Basics of WebRTC getStats() API
- 1. See blog post for code examples http://www.callstats.io/2015/07/06/basics-webrtc-getstats-api/ Some packets are lost on the way Some packets do not arrive in time Belated packets are discarded Decoder has to use incomplete data Black screen or pixelated image in video Audio may disappear Symptoms Core Metrics: Packet loss and discard Video: Loss of lip sync Audio: Elongated or cut-off syllables Symptoms Receiving interval change Receiving order change Sending order and interval 1 2 3 1 3 2 1 2 3 time Core Metrics: Jitter internet audio renderer Receive TCP or UDP audio de- packetizer audio decoder video decoder video de- packetizer video renderer 4. Receiver media render statistics: corresponds to the media rendering, typically frames lost, frames discarded, frames rendered, playout delay, etc. 3. Receiver RTP statistics: corresponds to the media receiver, typically packets received, bytes received, packets discarded, packets lost, jitter, etc audio source audio encoder audio packetizer video packetizer video encoder video source Send TCP or UDP 2. Sender RTP statistics: corresponds to the media sender, typically packets sent, bytes sent, round-trip-time, etc. 1. Sender media capture statistics: corresponds to the media generation, typically frame rate, frame size, clock rate of the media source, the name of the codec, etc. Media Flow and getStats() Structure OR webrtc-internals page* getStats() API call Accessing the statistics delays lost packets connection disruptions Network congestion is common on the Internet and it causes, for example, ... ...and that is why the WebRTC standard includes a statistics API. Web RTC is an INFOGRAPHIC: BASICS OF WEBRTC GETSTATS() API * only chrome and opera