Latency is the time it takes for a process to complete.
In WebRTC this can be referred to many different tasks within the media path.
The lower the latency the better the perceived media quality will be.
Latency can be measured in many areas:
- The time it takes to acquire a media frame from the microphone or the camera
- The time it takes to encode or decode a media frame
- The time it takes the network to send the packets, including any relay points along the way
- The time it takes the Jitter Buffer to process packets until it sends them to playback
- The time it takes for the display to show the video frame sent to it
