Hi HN, I'm building a browser MMO and I face the classic issue everybody faced: paquet order.
Currently I'm using websocket over TCP but I'm not satisfied with this solution as a packet lost means the stream will be interrupted.
There is the webRTC option with unordered and unreliable settings but it seems a real pain to set it up, it looks pretty unstable from what I could ear from users and it seems to have an amount of data transmited which is over what websocket transmits.
Finally the http3 standard seems to be able to answer to my issue as it uses udp and not tcp. But http3 does implement the blocking packet order in each multiplexed stream (which is expected for a classic web data transfer).
Does anyone know if it's possible to use some webRTC unreliable/unordered setting like to http3? Does the spec offer a way to have real UDP connection without the ordering overload?
Finally the http3 standard seems to be able to answer to my issue as it uses udp and not tcp. But http3 does implement the blocking packet order in each multiplexed stream (which is expected for a classic web data transfer).
Does anyone know if it's possible to use some webRTC unreliable/unordered setting like to http3? Does the spec offer a way to have real UDP connection without the ordering overload?