Sets the video buffer
This feature will increase the size of the audio and video playout delay by means of tweaking the webRTC jitter buffer pipeline (or a related buffer).
This can effectively be used as a way to delay the incoming video and audio by upwards of around 15-seconds. It's compatible with modern Chromium-based browsers.
While in theory this option can also help to improve video and audio quality, as a larger playback buffer can help reduce the effects of network jitter and packet loss, it's not a miracle solution in this regard. Adding 200-ms of buffer delay using this feature is worth trying however, as some users have reported it has helped improve their connections.
While one might think adding 10-seconds of buffer would then only improve the connection further, at this point it doesn't really. Work is being done to change this however, such as the work related to the
&chunkedtransfer mode, which will work quite well with extended buffer times.
&buffer=0will force the audio to be in sync with the video, with the video playing back with minimal delay.
&buffer=100will add a 100-ms time delay to the video, on top of any existing delay.
&buffer=200can help reduce video problems, such as frame jitter, with 200-ms of added delay.