Comparing with the google webrtc implementation for swift, the following appear to be missing from collectable statistics:

• media-source:
  • framesPerSecond
• outbound-rtp
  • qualityLimitationReason
  • totalPacketSendDelay
  • framesSent

It would be nice to have these in order to implement backpressure detection for video tracks and drop frames on the side of the sender.
Otherwise I am running into the issue of clients becoming progressively more and more behind on video tracks if network latency is very high.

I'm working around it, kind of, right now by checking remote-inbound-rtp's round trip time for spikes, but this is less effective.