Dropping video quality in low upload bandwidth network

[Dinith Minura]

Hi Everyone,

I'm new to WebRTC and I need to know about how video quality handling happen in a low upload bandwidth network environment. I have search for this problem over the internet but couldn't find a proper explanation on this. 

As I know, since WebRTC use UDP for video transferring and dropping data packets in a low bandwidth environment. Lets assume we start to stream a video with data packets like A,B,C,D,E ... is that mean in a low bandwidth network is it going to drop data packet "C" and send A,B,D,E .. or is it going to compress those data with an algorithm and send all data packets with the initial order? Can someone explain this scenario to me or send a link or source which explain these please.

   

[Sergio Garcia Murillo]

libwebrtc implements a congestion control to detect available bandwidth and adapt codec output appropriately.

Check Gustavo's great blog post for further information:

 http://www.rtcbits.com/2017/01/bandwidth-estimation-in-webrtc-and-new.html

Best regards
Sergio

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