In the past, we tried to bypass the encoding part of the WebRTC project with little luck (we wanted to transfer data that had already been encoded for several WebRTC clients). I got the impression that it is very closely integrated with the quality of service. WebRTC wants to configure encoder settings based on current network traffic.
The best solution we found was to actually run our own WebRTC using the dtlssrtpenc , nicesink and nicesrc in the OpenWebRTC project:
https://github.com/EricssonResearch/openwebrtc-gst-plugins
It was not at all easy to do. WebRTC has a very complicated handshake, and these GStreamer elements require a large number of special connections, but this gave the desired results.
Oh, and by the way, our experience is that openh264 works well for WebRTC traffic, and we ended up using it for many cases.
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