The way you drop latency is by reducing or eliminating buffers, that's pretty much it.

There's a trade-off with h264 and hevc/265, as with most video codecs - the larger the buffer you have, the more delta frames you can have between key frames.

This means lower bandwidth and better compression.

So it's a constant struggle between compression and latency.

The lowest latency that webrtc can get is about 4ms one way, 8ms round trip, at least based on the project I started to do realtime steaming with https://github.com/3DStreamingToolkit/3DStreamingToolkit

We proved you could do round trip times in the real world from cloud to customer in under 25ms.

But it comes with a lot of tradeoffs and cost and complexity to make it work.

Google Stadia fundamentally uses webrtc under the covers, although they use QUIC instead of TCP/ICE.

If you ripped out all of the congestion management from the video engine in webrtc, you'd reduce another several ms roundtrip at the cost of basically no network resilience.