Native WebRTC dropping frames

Question

Summary: How do I stream high quality video using WebRTC native?

I have an h264 stream that's 1920x1080 at about 30fps. I can currently stream this from a server on localhost to a native client on localhost just fine.

I wrote a WebRTC server using Google's WebRTC native library. I've written a VideoEncoder and VideoEncoderFactory that takes frames consisting of already encoded data and and broadcasts it over a video track. Using this I can send my h264 stream to the WebRTC server over a pipe and I can see the video stream in a browser.

However, any time something moves the video gets corrupted. It continues to play but is full of artifacts. Eventually I discovered that WebRTC is dropping some of my frames. When I attach a sequentially increasing ID to each frame before I pass it to rtc::AdaptedVideoTrackSource::OnFrame and I log this same ID in webrtc::VideoEncoder::Encode I can see that some of my frames simply disappear.

This kind of makes sense, I'm trying to stream high quality video over something meant for video chat and lowing my framerate fixes the corruption. However, I'm not asking the WebRTC library to do a lot, it's just forwarding already encoded data to a client on localhost. I have a native app that does this fine and I've seen one browser WebRTC client that can do this. Is there a field in the SDP or some configuration change that will allow me to stream my video?

Answer 1

This was the solution How to control bandwidth in WebRTC video call? .

I had heard about changing the offer sdp but dismissed it because I was told that the browser will accept unlimited bandwidth by default and that you'd only need to to this if you want to limit bandwidth. However, adding "b=AS:high number" has fixed all of my problems.