To make sure WebRTC conferences can be offered at the best possible quality, the WebRTC standard includes a statistics API. The statistics can be retrieved with the getStats() API call.

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