Quic already does this end to end.
The only use case I can think of might be gaming. If you had a HTTP/Websockets based game you wanted to cut latency down on you could have the FF proxy hosted near the game server.
Quic already does this end to end.
The only use case I can think of might be gaming. If you had a HTTP/Websockets based game you wanted to cut latency down on you could have the FF proxy hosted near the game server.
I can tell you a real life use case that already exists because I built this exact thing already (though a closed-source version): satellite internet terminals.
With satellite internet, uplink (upload) is generally more complicated than downlink (download). This is because a terminal can just passively listen for packets intended for him, but with uplink you have to make sure you only transmit in your designated time slot.
Because uplink is more complicated than downlink, more often than not uplink will get in a degraded or broken state, while downlink is still operational.
Now say I get a call from a customer service rep saying a customer's terminal is stuck in a degraded state. How can I recover it? TCP (and therefore HTTP) is a no-go because it requires uplink to be fully functional. So I can't visit the terminal's embedded webserver or issue commands to the HTTP API.
However, using a udp to tcp proxy like this, I can issue commands to the terminal's HTTP API "blindly" which it will then execute to (hopefully) recover itself. This can also be done with other "one-way" protocols like SNMP.
Unless it's a mass issue across a whole beam, it usually requires some investigation to figure out why an individual terminal state is degraded. And sometimes (more than I'm comfortable with) during the course of your investigation you make the terminal state even worse (because you tweaked some config parameter the wrong way). You can always tell when you done messed up because suddenly you get disconnected from your ssh session or you can no longer reach the embedded web server from your browser. In those cases sometimes the only way to recover is to precisely undo what you did, and an auto-ping-reset will not fix it (unless the system keeps track of most recent config changes or something). Your options are: use udp/tcp proxy or snmp to attempt to undo the config parameter you foobared or send out a technician and fix it on site...
I'm not saying our system is perfect. Surely we could improve fault detection and recovery. I'm just saying one-way communication protocols are real nice to have sometimes because of these scenarios.
( yes, I know that's a terrestrial, microwave relay dish and not a satellite antenna, but it could happen.)
If you handle the SDP and SCTP handshake you can force it to only use UDP via SCTP. Chrome and FF both support it today.
It's just -- and this may be mildly out of date -- EXTREMELY hard to find a server-side setup that will let you do all of what the browsers now support with WebRTC.
My recollection from the last time I skipped across this topic was: there are 1 or 2 heavyweight open source projects that mostly focus on the video aspects but also supported data channel stuff, and then one lightweight C and one lightweight Go project that supported everything, and you were mostly out of luck otherwise.
I remember the last time I checked, a year ago or so, pieces of what's required being worked on in recent OTP releases in BEAM-land. Hmmm ... time to check again.
I don't mean to sound smug, but I'm the author of the Jackbox Games multiplayer servers, I regularly handle hundreds of thousands of simultaneous websocket connections from players' phones. I can't for the life of me find a server implementation of webrtc data channels that I would actually put into production. I've seen a few blog posts with toy examples, but nothing that is ready for production. It sounds nice in theory, but try actually finding a production-ready server implementation of WebRTC data channels. I would LOVE to find a documented example of doing client-server communication with WebRTC data channels (a Go implementation on the server specifically). I've found no such thing. Every few months I search and don't find such a thing, and I don't need the extra few ms of latency improvement badly enough that it would be worth the engineering time to author it myself. If you know of one that's actually ready for primetime, I'd love to hear about it.
edit: after writing this comment, I did some snooping around since I haven't snooped around on this topic in a few months, this looks promising, unclear how widely used people are using it for production work: https://github.com/pion/webrtc
I got it working far enough to handle the WebRTC handshake over SCTP and get some basic data flowing. It's still in pretty rough shape and ran out of time to document it to any degree where someone else could take it and run with it.
If there's enough interest I might pick it up again but right now I've got other priorities going on.
I'm actually a little surprised the Jackbox games aren't monetized differently. They cost so little up front and that CPU time can't be free.
Just today I set up my wife's laptop with Drawful 2 and Jackbox 6 so she can play with her coworkers tomorrow. Thanks!
I have, but I didn't use go. I wrote a webrtc server in C, and probably handled something north of a trillion event points per month this way (receiving telemetry from interactive digital advertisements).
My goal wasn't latency but capacity: I can handle maybe 100kqps in HTTP on the same kit that could handle something like 600kqps+ messages with this (Never did proper limit testing here, since I knew my supplier was limited on the network links). It took about three weeks to get operational to a point it could be added to production, which was worth it to me.