grpc-web VS webtransport

* Additionally, client can stream data to the backend server (if bidirectional GRPC streams are used). I.e. client sends WebSocket messages, those will be transformed to GRPC messages by WebSocket server and delivered to the application backend.

As a result we have a system which allows to quickly create individual streams by using strict GRPC contract but terminating connections over WebSocket transport. So it works well in web browsers. After that no need to write WebSocket protocol, client implementation, handle WebSocket connection. This all will be solved by a suggested WebSocket server and its client SDKs.

The mechanics is similar to Websocketd (https://github.com/joewalnes/websocketd), but instead of creating OS processes we create GRPC streams. The difference from grpc-web (https://github.com/grpc/grpc-web) is that we provide streaming capabilities but not exposing GRPC contract to the client - just allowing to stream any data as payload (both binary and text) with some wrappers from our client SDKs side for managing subscriptions. I.e. it's not native GRPC streams on the client side - we expose just Connection/Subscription object to stream in both directions. GRPC streams used only for communication between WebSocket server and backend. To mention - grpc-web does not support all kinds of streaming now (https://github.com/grpc/grpc-web#streaming-support) while proposed solution can. This all should provide a cross-platform way to quickly write streaming apps due to client SDKs and language-agnostic nature of GRPC.

I personally see both pros and cons in this scheme (without concentrating on both too much here to keep the question short). I spent some time thinking about this myself, already have some working prototypes – but turned out need more opinions before moving forward with the idea and releasing this, kinda lost in doubts.

My main question - whether this seems interesting for someone here? Do you find this useful and see practical value?

By default, web browsers do not support gRPC, but we will use gRPC-web to make it possible.

You just have to use a library implementation for JavaScript https://github.com/grpc/grpc-web

TypeScript support remains an experimental feature of gRPC.

The gRPC-Web protocol supports HTTP/1 and can be used from a browser.

-- grpc-web

gRPC is mainly used in server-to-server communication, but it can also be used in client-to-server communication. gRPC-web is a gRPC implementation for web browsers. It is a JavaScript library that allows you to call gRPC services from a web browser. It supports Unary and Streaming Server API calls.

Since we're using Envoy, there's one more neat trick that we can employ. It turns out that Envoy also support gRPC-Web out of the box, a JavaScript client designed to support gRPC communication from the browser! That means that we can send gRPC messages over HTTP/1.1 as base64 encoded strings or as binary protobufs. Messages will be sent through our proxy and on to our backend service. The advantage of this is smaller and more efficient wire communication which should lead to better performance.

protoc-gen-grpc-web — a plugin that allows our front end to communicate with the backend using gRPC calls. A separate blog post on this coming up in the future.

WebTransport is now enabled by default and will be going to release with 114. As the original Explainer notes, it enables multiple use-cases that are hard or impossible to handle without it, especially for Gaming and live streaming. It covers cases that are problematic for alternative mechanisms, such as WebSockets. Built on top of HTTP3 (HTTP2 support will be coming later). The current implementation in Firefox is passing 505 out of 565 Web-Platform Tests.

WebTransport is still an emerging technology. As of November 2022, WebTransport is a draft specification with W3C, and there’s always a chance that aspects related to how it works may change.

Sort of. The browser will re-use the connection if you have a bunch of resources in the HTML. When rendering it sees that it needs 2 images and 3 javascript files from the same server, so it pipelines all of those. But for requests initiated from javascript, you're going to get a new connection for each one unless you're using a library that implements the long-polling hack. SocketIO can use the long-polling hack as a fallback if websockets is not supported. HTTP/2 (formerly SPDY) gets part of the way to replacing websockets, but it's not a synchronous link. Only the client can send messages to the server and the server can only respond to those message (with websockets, either side can send messages once the connection is open). FWIW, less than 50% of websites use HTTP/2. HTTP/3's webtransport looks like it could replace websockets, but it also looks like it'll live along side websockets.

If it's streaming data like dashboard statistics the new WebTransport API might be a much better base: https://github.com/w3c/webtransport/blob/main/explainer.md

Are you guys looking into the Web Transport protocol for the future? Right now you have to tunnel the websocket connections over http2 and it will probably be the same for http3 afaik.

I know there is this work in progress (https://w3c.github.io/webtransport/) and websockets are probably fine for a long time but sooner or later (unless there is an update to websockets) it will probably be faster to just do normal http requests and listen on server sent events.

What are your thoughts for Liveview for the future? Will it forever stay on websockets or would you be open to change the underlying technology if / when new stuff becomes available?

The W3C draft is here: https://github.com/w3c/webtransport

It's taken nearly 10 years for QUIC to be refined and adopted in the wild and we're basically there. There's even a new browser API in the works called WebTransport.

tl;dr using WebRTC just for realtime client<->server data sucks, but WebTransport[1] is coming soon to serve that exact usecase with an easy API

WebRTC has data channels, which are currently the only way to achieve unreliable and unordered real-time communication (UDP-style) between the browser and other browsers or a server. This is pretty essential for any networked application where latency is critical, like voice and video and fast-paced multiplayer games.

As other commenters have noted, it's a royal pain in the ass to set up WebRTC if all you want is UDP-style communication between a server and browser, since you need to wrangle half a dozen other protocols in the process.

However! A new API, WebTransport[1], is actively being developed that will offer a WebSockets-like (read: super simple to set up) API for UDP-style communication. I am extremely excited about it and its potential for real-time browser-based multiplayer games (which I'm working on).

https://github.com/w3c/webtransport