Does Google webrtc native implementation have support for SFU?

Question

1. Does Google WebRTC Native implementation has support for SFU?
2. Does Google WebRTC Native implementation support for integrating custom/hardware encoder/decoder?

Answer 1

Not without alteration.

Internally WebRTC's internal audio/video pipelines are directly tied to encoder/decoders.

PeerConnectionFactory allows you to provide a video decoder/encoder factory, so you can short circuit the logic here, and grab the encoded frames, mock up a stream, and feed them directly into it as a relay, creating a new PeerConnection and setting those streams onto it.

The audio end is more difficult. There isn't a codec factory, so you will have to short circuit the logic there probably by alteration of libwebrtc.

The final question is RTCP termination, and how to override the mechanisms for quality/bandwidth control to not create a "One goes out, they all go out." situation.

Since libwebrtc will be the SFU, it will receive RTCP feedback from its remote peer for the content it is proxying, and vice versa.

For a 1-1 situation, it needs to be able to forward the RTCP feedback to the remote peer.

For multipoint, it needs to perform some logic to determine if one of the peers is problematic, and stop sending it video, switch off its video feed, or attempt to switch to a lower bitrate video stream. Basically it needs to act as a conduit that attempts to predict why/how packet loss is occurring, and keep as many audio/video feeds operating normally at at the highest possible quality for each peer.

How exactly to hijack the RTCP feedback mechanisms in libwebrtc, I think that again will likely require some customization/hooks into libwebrtc

Answer 2

I think it will be easier to try with GStreamer implementation of WebRTC. Although it is still in "Bad Plugins" it is way easier to get or provide encoded audio and video. Actually it is implemented in that in mind - to make implementation of MFU and SFU easier.