WebRTC is a great protocol for low-latent streaming, particularly in one-to-one or one-to-a-few workflows. However, there are only a few codecs that are supported in the prevalent WebRTC libraries out there, and the ones that are supported now haven't always been supported. A use case for security cameras arose well before H264 support was prevalent and so we needed to provide a way of peering with WebRTC while streaming content encoded as H264/AAC.
The talk will go into the details of the two different solutions we came up with for solving this problem: 1) for mobile: modify the library to extract RTP packets or raw frames at relevant stages and use external player libraries (VLC, Exo Player, etc) to do the actual decode and rendering 2) for browsers: use the datachannel mechanism to send FMP4 video/audio data and then proxy that into MSE, similar to how CMAF works now.
The talk would then look forward to how this could be applied to non-supported codecs like HEVC and as a bridge to native AV1 support.
If time allows the talk would finish with observations and lessons learned from doing this in production.
Presented at Demuxed 2019 in San Francisco.
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