问题
I have a use case that doesn't exactly require real time communication, but as close as I can get it to make sure my users don't have to wait forever uploading a file after the fact. Our site allows users to record video and audio that we then store on our server.
I moved from flash, because flash, to a WebRTC implementation with a Kurento media server. This works super well with high quality internet connections but is relatively lacking for users with poor connectivity.
I was thinking if I could some how implement a bit of a buffer before feeding the server so that I can make up for dips in connectivity that would be ideal. I understand this isn't what WebRTC was meant for which leads to the question of whether or not there is another solution I should look into?
Any and all suggestions are welcome and appreciated! Thanks!
回答1:
A buffer wont change WebRTC's realtime characteristics in the face of low bandwidth. With video for instance, WebRTC might degrade either frame rate or resolution to keep up, but no-one has proposed degrading delivery time (dropping realtime-ness). Yours is an interesting use case however.
You could use MediaRecorder. It would let you record locally and then send the data as blobs to the server with either websockets or data channels.
Here's a fiddle that works in Chrome (which does video only unfortunately) and Firefox:
https://jsfiddle.net/tmwLxjLy
It shows simple recording and then playback of the video 5 seconds later. I'll leave it as an exercise to modify it to cause a continuous 5-second delay.
来源：https://stackoverflow.com/questions/34972529/can-i-use-some-sort-of-local-storage-as-a-temporary-holding-place-for-getusermed