Can I use some local storage as a temporary holding for getUserMedia for the nearest RTC?

Question

I have a use case that definitely does not require real-time sharing, but as close to it as possible to make sure that my users do not have to wait forever to download the file after the fact. Our site allows users to record video and audio, which we then store on our server.

I switched from flash, as flash, to the WebRTC implementation with the Kurento media server. This works very well with high-quality Internet connections, but is relatively not enough for users with poor connection.

I was thinking if I could somehow implement a buffer bit before serving the server so that I could compensate for connectivity failures that would be ideal. I understand that this is not what WebRTC was intended for, why the question arises, is there another solution that I should pay attention to?

Any suggestions are welcome and appreciated! Thanks!

Answer 1

The buffer does not alter the real-time characteristics of WebRTC in the face of low bandwidth. For example, in a WebRTC video, it can degrade either the frame rate or the resolution in order to keep up, but no one suggested reducing the delivery time (real-time reduction). However, your interesting use case.

You can use MediaRecorder . This will allow you to record locally and then send data as drops to the server using webcams or data channels.

Here is a fiddle that works in Chrome (unfortunately this video) and Firefox:

https://jsfiddle.net/tmwLxjLy

Shows simple recording and then video playback after 5 seconds. I will leave it as an exercise to change it to cause a continuous 5 second delay.