HTML5 websockets currently use a form of TCP communication. However, for real-time games, TCP just won't cut it (and is great reason to use some other platform, like native). As I probably need UDP to continue a project, I'd like to know if the specs for HTML6 or whatever will support UDP?
Also, are there any reliable benchmarks for WebSockets that would compare the WS protocol to a low-level, direct socket protocol?
On a LAN, you can get Round-trip times for messages over WebSocket of 200 microsec (from browser JS to WebSocket server and back), which is similar to raw ICMP pings. On MAN, it's around 10ms, WAN (over residential ADSL to server in same country) around 30ms, and so on up to around 120-200ms via 3.5G. The point is: WebSocket does add virtually no latency to the one you will get anyway, based on the network.
The wire level overhead of WebSocket (compared to raw TCP) is between 2 octets (unmasked payload of length < 126 octets) and 14 octets (masked payload of length > 64k) per message (the former numbers assume the message is not fragmented into multiple WebSocket frames). Very low.
For a more detailed analysis of WebSocket wire-level overhead, please see this blog post - this includes analysis covering layers beyond WebSocket also.
More so: with a WebSocket implementation capable of streaming processing, you can (after the initial WebSocket handshake), start a single WebSocket message and frame in each direction and then send up to 2^63 octets with no overhead at all. Essentially this renders WebSocket a fancy prelude for raw TCP. Caveat: intermediaries may fragment the traffic at their own decision. However, if you run WSS (that is secure WS = TLS), no intermediaries can interfere, and there you are: raw TCP, with a HTTP compatible prelude (WS handshake).
WebRTC uses RTP (= UDP based) for media transport but needs a signaling channel in addition (which can be WebSocket i.e.). RTP is optimized for loss-tolerant real-time media transport. "Real-time games" often means transferring not media, but things like player positions. WebSocket will work for that.
Note: WebRTC transport can be over RTP or secured when over SRTP. See "RTP profiles" here.
I would recommend developing your game using WebSockets on a local wired network and then moving to the WebRTC Data Channel API once it is available. As #oberstet correctly notes, WebSocket average latencies are basically equivalent to raw TCP or UDP, especially on a local network, so it should be fine for you development phase. The WebRTC Data Channel API is designed to be very similar to WebSockets (once the connection is established) so it should be fairly simple to integrate once it is widely available.
Your question implies that UDP is probably what you want for a low latency game and there is truth to that. You may be aware of this already since you are writing a game, but for those that aren't, here is a quick primer on TCP vs UDP for real-time games:
TCP is an in-order, reliable transport mechanism and UDP is best-effort. TCP will deliver all the data that is sent and in the order that it was sent. UDP packets are sent as they arrive, may be out of order, and may have gaps (on a congested network, UDP packets are dropped before TCP packets). TCP sounds like a big improvement, and it is for most types of network traffic, but those features come at a cost: a delayed or dropped packet causes all the following packets to be delayed as well (to guarantee in-order delivery).
Real-time games generally can't tolerate the type of delays that can result from TCP sockets so they use UDP for most of the game traffic and have mechanisms to deal with dropped and out-of-order data (e.g. adding sequence numbers to the payload data). It's not such a big deal if you miss one position update of the enemy player because a couple of milliseconds later you will receive another position update (and probably won't even notice). But if you don't get position updates for 500ms and then suddenly get them all out once, that results in terrible game play.
All that said, on a local wired network, packets are almost never delayed or dropped and so TCP is perfectly fine as an initial development target. Once the WebRTC Data Channel API is available then you might consider moving to that. The current proposal has configurable reliability based on retries or timers.
Here are some references:
WebRTC Introduction
WebRTC FAQ
WebRTC Data Channel Proposal
To make a long story short, if you want to use TCP for multiplayer games, you need to use what we call adaptive streaming techniques. In other words, you need to make sure that the amount of real-time data sent to synchronize the game world among the clients is governed by the currently available bandwidth and latency for each client.
Dynamic throttling, conflation, delta delivery, and other mechanisms are adaptive streaming techniques, which don't magically make TCP as efficient as UDP, but make it usable enough for several types of games.
I tried to explain these techniques in an article: Optimizing Multiplayer 3D Game Synchronization Over the Web (http://blog.lightstreamer.com/2013/10/optimizing-multiplayer-3d-game.html).
I also gave a talk on this topic last month at HTML5 Developer Conference in San Francisco. The video has just been made available on YouTube: http://www.youtube.com/watch?v=cSEx3mhsoHg
There's no UDP support for Websockets (there really should be), however you can apparently use WebRTC's RTCDataChannel API for UDP-like communication. There's a good article here:
http://www.html5rocks.com/en/tutorials/webrtc/datachannels/
RTCDataChannel actually uses SCTP which has configurable reliability and ordered delivery. You can get it to act like UDP by telling it to deliver messages unordered, and setting the maximum number of retransmits to 0.
I haven't tried any of this though.
I'd like to know if the specs for HTML6 or whatever will support UDP?
WebSockets won't. One of the benefits of WebSockets is that it piggybacks the existing HTTP connection. This means that to proxies and firewalls WebSockets looks like HTTP so they don't get blocked.
It's likely arbitrary UDP connections will never be part of any web specification because of security concerns. The closest thing to what you're after will likely come as part of WebRTC and it's associated JSEP protocol.
are there any reliable benchmarks ... that .. compare the WS protocol to a low-level, direct socket protocol?
Not that I'm aware of. I'm going to go out on a limb and predict WebSockets will be slower ;)
Related
As far as i know, TCP break down a message into segments. So, Why is multiplexing again on HTTP2? What are the benefits of multiplexing twice?
TCP isn’t multiplexed. TCP is just a guaranteed messaging stream (i.e. missing packets are re-requested and the TCP stream is basically temporarily blocked while this happens).
TCP, as a packet based protocol, can be used for multiplexed connections if the higher level application protocol (e.g. HTTP) allows sending of multiple messages. Unfortunately HTTP/1.1 does not allow this: once a HTTP/1.1 message is sent, no other message can be sent on that connection until that message is returned in full (ignoring the badly supported pipelining concept). This means HTTP/1.1 is basically synchronous and, if the full bandwidth is not used and other HTTP messages are queued, then it wastes any extra capacity that could be used on the underlying TCP connection.
To get around this more TCP connections can be opened, which basically allows HTTP/1.1 to act like a (limited) multiplexed protocol. If the network bandwidth was fully utilised then those extra connections would not add any benefit - it’s the fact there is capacity and that the other TCP connections are not being fully utilised that means this makes sense.
So HTTP/2 adds multiplexing to the protocol to allow a single TCP connection to be used for multiple in flight HTTP requests.
It does this by changing the text-based HTTP/1.1 protocol to a binary, packet-based protocol. These may look like TCP packets but that’s not really relevant (in the same way that saying TCP is similar to IP because it’s packet based is not relevant). Splitting messages into packets is really the only way of allowing multiple messages to be in flight at the same time.
HTTP/2 also adds the concept of streams so that packets can belong to different requests - TCP has no such concept - and this is what really makes HTTP/2 multiplexed.
In fact, because TCP doesn’t allow separate, independents streams (i.e. multiplexing), and because it is guaranteed, this actually introduces a new problem where a single dropped TCP packet holds up all the HTTP/2 streams on that connection, despite the fact that only one stream should really be affected and the other streams should be able to carry on despite this. This can even make HTTP/2 slower in certain conditions. Google is experimenting with moving away from TCP to QUIC to address this.
More details on what multiplexing means under HTTP/2 (and why it is a good improvement!) in my answer here: What does multiplexing mean in HTTP/2
TCP doesn't do multiplexing. The TCP segments just means that the (single) stream data is chopped up into pieces that can be sent in IP packets. Each TCP segment is only identified with a stream offset (sequence number), not with any useful way to identify separate streams. (We'll ignore the rarely-useful Urgent Pointer thing.)
So to do multiplexing, you need to put something on top of TCP. Which HTTP/2 does.
HTTP & HTTP/2 are both application level protocols that must utilize a lower level protocol like TCP to actually talk on the Internet. The protocol of the Internet is generally TCP over IP over Ethernet.
It looks like this:
As you can see HTTP is sitting above TCP. Below TCP is IP. One of the main protocols of the Internet. IP itself deals with packets which are switched/multiplexed. I think that's where you might be getting the idea that TCP is multiplexed, it's not. Think of a TCP connection as being like a single lane road tunnel where no one can pass. Lets say it has one single lane in each direction. This is what a TCP connection would look like. A tunnel where you put data in one end, and it comes out the other in the same order it went in. That is TCP. You can see there is no multiplexing on that. However, TCP does provides a reliable connection protocol for which other protocols may be built on top of like HTTP. And reliability is essential for HTTP.
HTTP 1.1 is simply a request response protocol. But as you know, it's not multiplexed. So only allow one outstanding request at a time and has to send the whole response to each request at a time. Previously the browsers got around that limitation by creating multiple TCP connections (tunnels) to the server with which to make more requests.
HTTP 2 actually splits the data up again and allows multiplexing over the one connection so that no further connections need to be created. It means the server can start servicing multiple requests and multiplex the responses so that the browser can start receiving images, pages and other resources at the same time, not one at a time.
Hope that makes it clear.
I have recently come across a term 'Reactive sockets'. Up until this point, I used to think websockets are the way to go for full fledged asynchronous style.
So what are reactive sockets.
This link (http://rsocket.io/) even talks about comparison over websockets.
What is RSocket?
RSocket implements the Reactive Streams specification over the network boundary. It is an application-level communication protocol with framing, session resumption, and backpressure built-in that works over the network.
RSocket is transport agnostic. RSocket can be run over Websockets, TCP, HTTP/2, and Aeron.
How does RSocket differ from Websockets?
Websockets do not provide application-level backpressure, only TCP-based byte-level backpressure. Websockets also only provide framing they do not provide application semantics. It is up to the developer to build out an application protocol for interacting with the websocket.
RSocket provides framing, application semantics, application-level backpressure, and it is not tied to a specific transport.
For more information on the motivations behind the creation of RSocket checkout the motivations doc on the RSocket site.
Both WebSocket and RSocket are protocols which feature bi-directional, multiplex, duplex communication. But both work at different levels.
WebSocket is a low level communication protocol layered over TCP. It defines how a stream of bytes is transformed into frames. But WebSocket message itself does not have instructions about how to route or process it. Therefore, we need messaging protocols that operate on top of websocket, at application level, to achieve two-way communication.
RSocket is a fully reactive application level protocol which runs over byte stream transports such as TCP, WebSocket, UDP or other. It provides application flow control over the network to prevent outages and increase resiliency. RSocket employs the idea of asynchronous stream processing with non-blocking back-pressure, in which a failing component will, rather than simply dropping traffic, communicate its stress to upstream components, getting them to reduce the load.
Websocket
TLDR: L4 protocol, TCP for web.
Websocket is single bytestream, frames based protocol with very compact header.
It relies on web/http in each important protocol aspect: http based handshake (full roundtrip, not ideal for latency), text frames in addition to binary ones, compression support (for low-throughput/high latency connections),
mandatory frame content masking for compatibility with legacy non-tls http proxies.
Frames may be fragmented for better memory utilization by client and server;
Flow control is byte level only from TCP, and is not propagated to userspace.
Because websocket is just single bytestream transport, It needs full application protocol on top to be useful, and application level flow control scheme to be scalable.
Adoption wise, stable websocket implementation is available for most OSes / architectures, protocol is supported by all browsers and is go-to solution if any traffic needs to survive internet hop.
RSocket. Theory
TLDR: L5 protocol, primarily cloud/datacenter communications with excellent
throughput/latency characteristic: huge throughput while maintaining latency < few millis.
RSocket is session layer protocol, offers multiplexed flow controlled streams of binary messages over any transport capable of transferring bytes in order (tcp, unix sockets, also websocket).
Low latency is cornerstone, protocol has several capabilities for this:
2 levels of flow control: Reactive-Streams on individual stream level,
request leasing on connection level. Leasing is feature to control number of active streams by responder side using service & connection latency stats.
Instant handshake: client may send requests immediately after initial setup message.
Message fragmentation helps with reducing server memory pressure and improves latency for large messages (if done properly, see RSocket. Practice below).
Session resumption: reduced latency on client reconnection.
Because binary streaming interactions / multiplexing are available out-of-the-box, It is trivial to implement application RPC on top - only data serialization/deserialization is needed (mstreams-rpc using protobuf data encoding).
The protocol is semantically compatible with http2, which means It is also compatible with GRPC (given protobuf is used for message encoding).
RSocket. Practice
Only useful on JVM because that's where reactive streams are popular and practically useful with several
stable implementations: rxjava, project-reactor, smallrye-mutiny.
RSocket/RSocket-java is based on project-reactor from springboot.
Natural expectation would be best-in-class throughput, unfortunately RSocket/RSocket-java did not get this
right so performs worse
than 10+ year older GRPC (its predecessor Stubby was in use from ~2001) on top of http2: chatty web protocol.
Fragmentation: no server memory use or latency improvement because RSocket/RSocket-java
implemented It in pointless way - frames are always reassembled before passing
downstream.
GRPC compatibility: absent.
Advice for 2022: better stick with GRPC.
I'm learning about HTTP/2 protocol. It's a binary protocol with small message frames. It allows stream multiplexing over single TCP connection. Conceptually it seems very similar to WebSockets.
Are there plans to obsolete websockets and replace them with some kind of headerless HTTP/2 requests and server-initiated push messages? Or will WebSockets complement HTTP/2?
After just getting finished reading RFC 7540, HTTP/2 does obsolete websockets for all use cases except for pushing binary data from the server to a JS webclient. HTTP/2 fully supports binary bidi streaming (read on), but browser JS doesn't have an API for consuming binary data frames and AFAIK such an API is not planned.
For every other application of bidi streaming, HTTP/2 is as good or better than websockets, because (1) the spec does more work for you, and (2) in many cases it allows fewer TCP connections to be opened to an origin.
PUSH_PROMISE (colloquially known as server push) is not the issue here. That's just a performance optimization.
The main use case for Websockets in a browser is to enable bidirectional streaming of data. So, I think the OP's question becomes whether HTTP/2 does a better job of enabling bidirectional streaming in the browser, and I think that yes, it does.
First of all, it is bi-di. Just read the introduction to the streams section:
A "stream" is an independent, bidirectional sequence of frames
exchanged between the client and server within an HTTP/2 connection.
Streams have several important characteristics:
A single HTTP/2 connection can contain multiple concurrently open
streams, with either endpoint interleaving frames from multiple
streams.
Streams can be established and used unilaterally or shared by
either the client or server.
Streams can be closed by either endpoint.
Articles like this (linked in another answer) are wrong about this aspect of HTTP/2. They say it's not bidi. Look, there is one thing that can't happen with HTTP/2: After the connection is opened, the server can't initiate a regular stream, only a push stream. But once the client opens a stream by sending a request, both sides can send DATA frames across a persistent socket at any time - full bidi.
That's not much different from websockets: the client has to initiate a websocket upgrade request before the server can send data across, too.
The biggest difference is that, unlike websockets, HTTP/2 defines its own multiplexing semantics: how streams get identifiers and how frames carry the id of the stream they're on. HTTP/2 also defines flow control semantics for prioritizing streams. This is important in most real-world applications of bidi.
(That wrong article also says that the Websocket standard has multiplexing. No, it doesn't. It's not really hard to find that out, just open the Websocket RFC 6455 and press ⌘-F, and type "multiplex". After you read
The protocol is intended to be extensible; future versions will likely introduce additional concepts such as multiplexing.
You will find that there is 2013 draft extension for Websocket multiplexing. But I don't know which browsers, if any, support that. I wouldn't try to build my SPA webapp on the back of that extension, especially with HTTP/2 coming, the support may never arrive).
Multiplexing is exactly the kind of thing that you normally have to do yourself whenever you open up a websocket for bidi, say, to power a reactively updating single page app. I'm glad it's in the HTTP/2 spec, taken care of once and for all.
If you want to know what HTTP/2 can do, just look at gRPC. gRPC is implemented across HTTP/2. Look specifically at the half and full duplex streaming options that gRPC offers. (Note that gRPC doesn't currently work in browsers, but that is actually because browsers (1) don't expose the HTTP/2 frame to the client javascript, and (2) don't generally support Trailers, which are used in the gRPC spec.)
Where might websockets still have a place? The big one is server->browser pushed binary data. HTTP/2 does allow server->browser pushed binary data, but it isn't exposed in browser JS. For applications like pushing audio and video frames, this is a reason to use websockets.
Edit: Jan 17 2020
Over time this answer has gradually risen up to the top (which is good, because this answer is more-or-less correct). However there are still occasional comments saying that it is not correct for various reasons, usually related to some confusion about PUSH_PROMISE or how to actually consume message-oriented server -> client push in a single page app.
If you need to build a real-time chat app, let's say, where you need to broadcast new chat messages to all the clients in the chat room that have open connections, you can (and probably should) do this without websockets.
You would use Server-Sent Events to push messages down and the Fetch api to send requests up. Server-Sent Events (SSE) is a little-known but well supported API that exposes a message-oriented server-to-client stream. Although it doesn't look like it to the client JavaScript, under the hood your browser (if it supports HTTP/2) will reuse a single TCP connection to multiplex all of those messages. There is no efficiency loss and in fact it's a gain over websockets because all the other requests on your page are also sharing that same TCP connection. Need multiple streams? Open multiple EventSources! They'll be automatically multiplexed for you.
Besides being more resource efficient and having less initial latency than a websocket handshake, Server-Sent Events have the nice property that they automatically fall back and work over HTTP/1.1. But when you have an HTTP/2 connection they work incredibly well.
Here's a good article with a real-world example of accomplishing the reactively-updating SPA.
From what I understood HTTP/2 is not a replacement for websocket but aims to standardize SPDY protocol.
In HTTP/2, server-push is used behind the scene to improve resource loading by the client from the browser. As a developer, you don't really care about it during your development. However, with Websocket, the developer is allowed to use API which is able to consume and push message with an unique full-duplex connection.
These are not the same things, and they should complement each other.
I say Nay (Websockets aren't obsolete).
The first and most often ignored issue is that HTTP/2 push isn't enforceable and might be ignored by proxies, routers, other intermediaries or even the browser.
i.e. (from the HTTP2 draft):
An intermediary can receive pushes from the server and choose not to forward them on to the client. In other words, how to make use of the pushed information is up to that intermediary. Equally, the intermediary might choose to make additional pushes to the client, without any action taken by the server.
Hence, HTTP/2 Push can't replace WebSockets.
Also, HTTP/2 connections do close after a while.
It's true that the standard states that:
HTTP/2 connections are persistent. For best performance, it is expected that clients will not close connections until it is determined that no further communication with a server is necessary (for example, when a user navigates away from a particular web page) or until the server closes the connection.
But...
Servers are encouraged to maintain open connections for as long as possible but are permitted to terminate idle connections if necessary. When either endpoint chooses to close the transport-layer TCP connection, the terminating endpoint SHOULD first send a GOAWAY (Section 6.8) frame so that both endpoints can reliably determine whether previously sent frames have been processed and gracefully complete or terminate any necessary remaining tasks.
Even if the same connection allows for pushing content while it is open and even if HTTP/2 resolves some of the performance issues introduced by HTTP/1.1's 'keep-alive'... HTTP/2 connections aren't kept open indefinitely.
Nor can a webpage re-initiate an HTTP/2 connection once closed (unless we're back to long-pulling, that is).
EDIT (2017, two years later)
Implementations of HTTP/2 show that multiple browser tabs/windows share a single HTTP/2 connection, meaning that push will never know which tab / window it belongs to, eliminating the use of push as a replacement for Websockets.
EDIT (2020)
I'm not sure why people started downvoting the answer. If anything, the years since the answer was initially posted proved that HTTP/2 can't replace WebSockets and wasn't designed to do so.
Granted, HTTP/2 might be used to tunnel WebSocket connections, but these tunneled connections will still require the WebSocket protocol and they will effect the way the HTTP/2 container behaves.
The answer is no. The goal between the two are very different. There is even an RFC for WebSocket over HTTP/2 which allows you to make multiple WebSocket connections over a single HTTP/2 TCP pipe.
WS over HTTP/2 will be a resource conservation play by decreasing the time to open new connections and allowing for more communication channels without the added expense of more sockets, soft IRQs, and buffers.
https://datatracker.ietf.org/doc/html/draft-hirano-httpbis-websocket-over-http2-01
Well, to quote from this InfoQ article:
Well, the answer is clearly no, for a simple reason: As we have seen above, HTTP/2 introduces Server Push which enables the server to proactively send resources to the client cache. It does not, however, allow for pushing data down to the client application itself. Server pushes are only processed by the browser and do not pop up to the application code, meaning there is no API for the application to get notifications for those events.
And so HTTP2 push is really something between your browser and server, while Websockets really expose the APIs that can be used by both client (javascript, if its running on browser) and application code (running on server) for transferring real time data.
As of today, no.
HTTP/2, compared to HTTP, allows you to maintain a connection with a server. From there, you can have multiple streams of data at the same time. The intent is that you can push multiple things at the same time even without the client requesting it. For example, when a browser asks for a index.html, the server might want to also push index.css and index.js. The browser didn't ask for it, but the server might provide it without being asked because it can assume you're going to want in a few seconds.
This is faster than the HTTP/1 alternative of getting index.html, parsing it, discovering it needs index.js and index.css and then building 2 other requests for those files. HTTP/2 lets the server push data the client hasn't even asked for.
In that context, it's similar to WebSocket, but not really by design. WebSocket is supposed to allow a bi-directional communication similar to a TCP connection, or a serial connection. It's a socket where both communicate with each other. Also, the major difference is that you can send any arbitrary data packets in raw bytes, not encapsulated in HTTP protocol. The concepts of headers, paths, query strings only happen during the handshake, but WebSocket opens up a data stream.
The other difference is you get a lot more fine-tuned access to WebSocket in Javascript, whereas with HTTP, it's handled by the browser. All you get with HTTP is whatever you can fit in XHR/fetch(). That also means the browser will get to intercept and modify HTTP headers without you being able to control it (eg: Origin, Cookies, etc). Also, what HTTP/2 is able to push is sent to the browser. That means JS doesn't always (if ever) know things are being pushed. Again, it makes sense for index.css and index.js because the browser will cache it, but not so much for data packets.
It's really all in the name. HTTP stands for HyperText Transfer Protocol. We're geared around the concept of transferring assets. WebSocket is about building a socket connection where binary data gets passed around bidirectionally.
The one we're not really discussing is SSE (Server-Sent Events). Pushing data to the application (JS) isn't HTTP/2's intent, but it is for SSE. SSE gets really strengthened with HTTP/2. But it's a not a real replacement for WebSockets when what's important is the data itself, not the variable endpoints being reached. For each endpoint in with WebSocket a new data stream is created, but with SSE it's shared between the already existing HTTP/2 session.
Summarized here are the objectives for each:
HTTP - Respond to a request with one asset
HTTP/2 - Respond to a request with multiple assets
SSE - Respond with a unidirectional text (UTF-8) event stream
WebSocket - Create a bidirectional binary data stream
Message exchange and simple streaming(not audio, video streaming) can be done via both Http/2 multiplexing and WebSockets. So there is some overlap, but WebSockets have well established protocol, a lot of frameworks/APIs and less headers overhead.
Here is nice article about the topic.
No, WebSockets are not obsolete. However, HTTP/2 breaks websockets as defined for HTTP/1.1 (mostly by forbidding protocol updates using the Upgrade header). Which is why this rfc:
https://datatracker.ietf.org/doc/html/rfc8441
defines a websocket bootstrapping procedure for HTTP/2.
For the time being April 2020, HTTP/2 is not making WebSockets obsolete. The greatest advantage of WebSockets over HTTP2 is that
HTTP/2 works only on Browser Level not Application Level
Means that HTTP/2 does not offer any JS API like WebSockets to allow communication and transfer some kind of JSON or other data to server directly from Application (e.g. Website). So, as far as I believe, HTTP/2 will only make WebSockets obsolete if it starts offering API like WebSockets to talk to server. Till that it is just updated and faster version of HTTP 1.1.
No HTTP/2 does not make websockets obsolete, but SSE over HTTP/2 offers a viable alternative. The minor caveat is that SSE does not support unsolicited events from server to client (and neither does HTTP/2): i.e. the client has to explicitly subscribe by creating an EventSource instance specifying the event source endpoint. So you may have to slightly reorganise how the client arranges for events to be delivered - I can't think of a scenario where this is actually a technical barrier.
SSE works with HTTP/1.1. But HTTP/2 makes using SSE generally viable and competitive with websockets in terms of efficiency, instead of practically unusable in the case of HTTP/1.1. Firstly, HTTP/2 multiplexes many event source connections (or rather "streams" in HTTP/2 terms) onto a single TCP connection where as in HTTP/1.1 you'd need one connection for each. According to the HTTP/2 spec, millions of streams can be created per connection by default with the recommended (configurable) minimum being 100, where as browsers maybe severly limited in the number of TCP connections they can make. Second reason is efficiency: many streams in HTTP/2 is requires much less overhead than the many connections required in HTTP/1.1.
One final thing is, if you want to replace websockets with SSE your forgoing some of the tools / middlewares built on top of websockets. In particular I'm thinking of socket.io (which is how a lot of people actually use websockets), but I'm sure there is a ton more.
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So I'm looking to build a chat app that will allow video, audio, and text. I spent some time researching into Websockets and WebRTC to decide which to use. Since there are plenty of video and audio apps with WebRTC, this sounds like a reasonable choice, but are there other things I should consider?
Feel free to share your thoughts.
Things like:
Due to being new WebRTC is available only on some browsers, while WebSockets seems to be in more browsers.
Scalability - Websockets uses a server for session and WebRTC seems to be p2p.
Multiplexing/multiple chatrooms - Used in Google+ Hangouts, and I'm still viewing demo apps on how to implement.
Server - Websockets needs RedisSessionStore or RabbitMQ to scale across multiple machines.
WebRTC is designed for high-performance, high quality communication of video, audio and arbitrary data. In other words, for apps exactly like what you describe.
WebRTC apps need a service via which they can exchange network and media metadata, a process known as signaling. However, once signaling has taken place, video/audio/data is streamed directly between clients, avoiding the performance cost of streaming via an intermediary server.
WebSocket on the other hand is designed for bi-directional communication between client and server. It is possible to stream audio and video over WebSocket (see here for example), but the technology and APIs are not inherently designed for efficient, robust streaming in the way that WebRTC is.
As other replies have said, WebSocket can be used for signaling.
I maintain a list of WebRTC resources: strongly recommend you start by looking at the 2013 Google I/O presentation about WebRTC.
Websockets use TCP protocol.
WebRTC is mainly UDP.
Thus main reason of using WebRTC instead of Websocket is latency.
With websocket streaming you will have either high latency or choppy playback with low latency. With WebRTC you may achive low-latency and smooth playback which is crucial stuff for VoIP communications.
Just try to test these technology with a network loss, i.e. 2%. You will see high delays in the Websocket stream.
WebSockets:
Ratified IETF standard (6455) with support across all modern browsers and even legacy browsers using web-socket-js polyfill.
Uses HTTP compatible handshake and default ports making it much easier to use with existing firewall, proxy and web server infrastructure.
Much simpler browser API. Basically one constructor with a couple of callbacks.
Client/browser to server only.
Only supports reliable, in-order transport because it is built On TCP. This means packet drops can delay all subsequent packets.
WebRTC:
Just beginning to be supported by Chrome and Firefox. MS has proposed an incompatible variant. The DataChannel component is not yet compatible between Firefox and Chrome.
WebRTC is browser to browser in ideal circumstances but even then almost always requires a signaling server to setup the connections. The most common signaling server solutions right now use WebSockets.
Transport layer is configurable with application able to choose if connection is in-order and/or reliable.
Complex and multilayered browser API. There are JS libs to provide a simpler API but these are young and rapidly changing (just like WebRTC itself).
webRTC or websockets? Why not use both.
When building a video/audio/text chat, webRTC is definitely a good choice since it uses peer to peer technology and once the connection is up and running, you do not need to pass the communication via a server (unless using TURN).
When setting up the webRTC communication you have to involve some sort of signaling mechanism. Websockets could be a good choice here, but webRTC is the way to go for the video/audio/text info. Chat rooms is accomplished in the signaling.
But, as you mention, not every browser supports webRTC, so websockets can sometimes be a good fallback for those browsers.
Security is one aspect you missed.
With Websockets the data has to go via a central webserver which typically sees all the traffic and can access it.
With WebRTC the data is end-to-end encrypted and does not pass through a server (except sometimes TURN servers are needed, but they have no access to the body of the messages they forward).
Depending on your application this may or may not matter.
If you are sending large amounts of data, the saving in cloud bandwidth costs due to webRTC's P2P architecture may be worth considering too.
Comparing websocket and webrtc is unfair.
Websocket is based on top of TCP. Packet's boundary can be detected from header information of a websocket packet unlike tcp.
Typically, webrtc makes use of websocket. The signalling for webrtc is not defined, it is upto the service provider what kind of signalling he wants to use. It may be SIP, HTTP, JSON or any text / binary message.
The signalling messages can be send / received using websocket.
Webrtc is a part of peer to peer connection.
We all know that before creating peer to peer connection, it requires handshaking process to establish peer to peer connection.
And websockets play the role of handshaking process.
Websocket and WebRTC can be used together, Websocket as a signal channel of WebRTC, and webrtc is a video/audio/text channel, also WebRTC can be in UDP also in TURN relay, TURN relay support TCP HTTP also HTTPS.
Many projects use Websocket and WebRTC together.
I am trying to understand the difference between WebRTC and WebSockets so that I can better understand which scenario calls for what. I am curious about the broad idea of two parties (mainly web based, but potentially one being a dedicated server application) talking to each other.
Assumption:
Clearly in regards to ad-hoc networks, WebRTC wins as it natively supports the ICE protocol/method.
Questions:
Regarding direct communication between two known parties in-browser, if I am not relying on sending multimedia data, and I am only interested in sending integer data, does WebRTC give me any advantages over webSockets other than data encryption?
Regarding a dedicated server speaking to a browser based client, which platform gives me an advantage? I would need to code a WebRTC server (is this possible out of browser?), or I would need to code a WebSocket server (a quick google search makes me think this is possible).
There is one significant difference: WebSockets works via TCP, WebRTC works via UDP.
In fact, WebRTC is SRTP protocol with some additional features like STUN, ICE, DTLS etc. and internal VoIP features such as Adaptive Jitter Buffer, AEC, AGC etc.
So, WebSockets is designed for reliable communication. It is a good choice if you want to send any data that must be sent reliably.
When you use WebRTC, the transmitted stream is unreliable. Some packets can get lost in the network. It is bad if you send critical data, for example for financial processing, the same issue is ideally suitable when you send audio or video stream where some frames can be lost without any noticeable quality issues.
If you want to send data channel via WebRTC, you should have some forward error correction algorithm to restore data if a data frame was lost in the network.
WebRTC specifies media transport over RTP .. which can work P2P under certain circumstances. In any case to establish a webRTC session you will need a signaling protocol also .. and for that WebSocket is a likely choice. In other words: unless you want to stream real-time media, WebSocket is probably a better fit.
Question 1: Yes. The DataChannel part of WebRTC gives you advantages in this case, because it allows you to create a peer to peer channel between browsers to send and receive any raw data you want. Websockets forces you to use a server to connect both parties.
Question 2 Like I said in the previous response, Websockets are better if you want a server-client communication, and there are many implementations to do this (i.e. jWebSocket). To add support in a server to establish a connection with a WebRTC DataChannel, it may take you some days of life and health. :)