Websocket latenct
admin Staff asked 8 years ago
I’m testing the new websocket streaming feature and it seems the latency is about 4-5 seconds more than RTMP.
Can I adjust the player buffer somehow to get the sub-second latency as mentioned?
1 Answers
ic Staff answered 8 years ago
To decrease the latency when using the websocket HTML5 player, key frames need to be sent as frequent as possible. Ideally, at least once every second. For example, if the current framerate is 30 fps, then the keyframe needs to be sent out every 30 frames or less. This is similar to reducing the GOP (group of pictures) size/length to match the current framerate.
These settings can be adjusted on the encoder side.
These settings can be adjusted on the encoder side.