[Sean-Der]

Would you mind explaining more the issue you are having with WebRTC?

It sounds like you want playoutDelay (no latency) added by the receiver?

You could also use insertable streams and modify/drop frames with some arbitrary logic.

[Tigress8780]

I was able to achieve a latency of ~30ms with a custom Android client by dropping outdated frames, but I could not find out how to do this with WebRTC, since its buffering and rendering is entirely controlled by the browser with almost no tuning knobs.

WebRTC as a technology is also quite complex, providing less flexibility than a custom TCP or WebSocket stream. For example, to further reduce latency, it is possible to tunnel TCP traffic through USB with the help of Android's debugging interface, but this interface does not allow tunneling of UDP, nor does webrtc-rs allows listening on 127.0.0.1 (it is hardcoded to ignore the loopback interface).

Insertable streams looks interesting, but it seems that Firefox and Safari (especially on iOS, where I can not easily relese a native client) does not support it at all, while WebCodecs is at least experimental.

[Sean-Der]

Would it help if I added the ability to gather on 127.0.0.1 to webrtc-rs' SettingEngine? That exists in Pion, but landed after webrtc-rs was started.

[Tigress8780]

Does TCP actually works on webrtc-rs and Chrome? ADB does not support UDP at all. If it works, then having the ability to gather loopback interface is certainly a plus for my particular use case.

[Sean-Der]

Yea!

WebRTC supports TCP candidates, so webrtc-rs can do the same. If that unblocks you happy to implement. Getting people with interested use cases involved in the project helps a lot.

[Tigress8780]

I have opened an issue on GitHub [1], we can continue there.

[1] https://github.com/webrtc-rs/webrtc/issues/426