An XMMP server sends push notifications to a client behind a NAT using a public endpoint( IP + Port) supplied by NAT to client. But how long this endpoint is assigned to this specific client by NAT, what will happen if NAT assigns same endpoint to another client ? How this problem can be solved?
XMPP uses a standard TCP connection. NATs will keep the association for as long as the connection is alive (unless they are horribly broken).
Update: The last part of my statement could have been expanded a bit. Horribly broken NAT implementations do exist. Generally these are a small percentage, but many (most?) popular XMPP clients do ensure they send some kind of keepalive over idle connections.
There are three kinds of keepalive you can use I'll list them here in order of bandwidth/processing requirements:
TCP keepalives are a good lightweight option, especially as once they are enabled, they are automatically handled by the OS. How to enable them will depend on your language and framework, but at the lowest level, you need to enable the SO_KEEPALIVE option on the socket.
There are two problems with TCP keepalives. One is that you can't control them from your application (unless you write platform-specific code). The second problem is that some NAT implementations are so broken that they will ignore TCP keepalives too! But you're hopefully down to a very small percentage now.
So another option is whitespace keepalives. Since these involve data going across the stream, you should be safe from even the broken NATs that ignore keepalives.
Whitespace keepalives simply involve sending the space character (' ') across the XMPP stream at any time it is idle. XML and XMPP allow unlimited whitespace between elements, and it is simply ignored by the recipient.
Finally, you can use fully-fledged XMPP pings (XEP-0199). These involve ending an actual <iq/> 'get' stanza to the server, which then must reply. This ensures data flows in both directions, and should make even the most broken NAT implementations keep your connection alive.
Ok, I should mention that there is an even worse class of NAT. I have seen NATs that will simply 'forget' about your mapping for a range of reasons, including their mapping table being full, or just after a timer. There is nothing you can do to work around these, they don't work with any long-lived TCP connections. The best you could probably do at that point is use BOSH (essentially XMPP over HTTP).
Conclusion: If you are concerned that your application may run behind some of these devices, I suggest something like the following algorithm (exact times may be tweaked, but I recommend these as minimum values):
If you have not sent any data for 60s, send a single space character.
If you have not received any data for 120s, send an XMPP ping to your server.
If the server doesn't reply to the ping within a reasonable amount of time, reconnect.
Because the behaviour of broken NAT devices is beyond any standard protocol specification, it is naturally impossible to devise a perfect solution that will work with all of them, all of the time. You just have to accept that these are a small minority, and none of this matters for working NAT devices (though there are other kinds of network breakages that may make regular keepalives/pings a good idea, depending on the needs of your application).
The Solution is sending keep alive messages to maintain the NAT entry. XMPP whitespace is typically used. Send it eg every Ten minutes to preserve reachability of the nated client.
You have to keep in mind that NAT is no standardized technique. Thus there are different implementations. The provided RFCs in the comment above is from the BEHAVE working group.
Related
I'm trying to understand if ZeroMQ can connect pub or sub socket to non existing (yet) ip address. Will it automatically connect when this IP address will appear in the future?
Or should I check up existance first before connecting?
Is the behavior same for PUB and SUB sockets?
The answer is buried somewhat in the manual, here:
for most transports and socket types the connection is not performed immediately but as needed by ØMQ. Thus a successful call to zmq_connect() does not mean that the connection was or could actually be established. Because of this, for most transports and socket types the order in which a server socket is bound and a client socket is connected to it does not matter. The ZMQ_PAIR sockets are an exception, as they do not automatically reconnect to endpoints.
As that quote says, the order of binding and connecting does not matter. This is extremely useful, as you don't then have to worry about start-up order; the client will be quite happy waiting for a server to come online, able to run other things without blocking on the connect.
Other Things That Are Useful
The direction of bind/connect is independent of the pattern used on top; thus a PUB socket can be connected to a SUB socket that has been bound to an interface (whereas the other way round might feel more natural).
The other thing that I think a lot of people don't realise is that you can bind (or connect) sockets more than once, to different transports. So a PUB socket can quite happily send to SUB clients that are both local in-process threads, other processes on the same machine via ipc, and to clients on remote machines via tcp.
There are other things that you can do. If you use the ZMQ_FD option from here, you can get ZMQ_EVENT notifcations in some way or other (I can't remember the detail) which will tell you when the underlying connection has been successfully made. Using the file descriptor allows you to include that in a zmq_poll() (or some other reactor like epoll() or select()). You can also exploit the heartbeat functionality that a socket can have, which will tell you if the connection dies for some reason or other (e.g. crashed process at the other end, or network cable fallen out). Use of a reactor like zmq_poll(), epoll() or select() means that you can have a pure actor model event-driven system, with no need to routinely check up on status flags, etc.
Using these facilities in ZMQ allows for the making of very robust distributed applications/system that know when various bits of themselves have died, come back to life, taken a network-out holiday, etc. For example, just knowing that a link is dead perhaps means that a node in your distributed app changes its behaviour somehow to adapt to that.
As I simply understands HTTP2 is m:1 pattern where you put m logical connections into 1 TCP stream
Is it possible to do m:n pattern in http2?
m streams are demuxed into n connections for better reliability, because often one single TCP breaks all h2 hangs.
It would be possible, but in practice it is not done.
Browsers especially try hard to open just one connection to a domain, and even reuse the same connection for different subdomains if they can figure out that it resolves to the same IP address and same certificate.
Other clients may implement a m:n scheme (for example, Jetty 9.4.x HTTP/2 client does - disclaimer: I'm the maintainer).
The problem of choosing a good n may not be trivial, and it risks to go back to HTTP/1.1 6-8 TCP connections per domain.
Since each connection would be multiplexed anyway, the failure of a single HTTP/2 connection would be worse than the failure of a single HTTP/1.1 connection (because it would fail multiple requests rather than just one), so I guess that it would not make that much of a difference with respect to a single HTTP/2 connection.
Google's QUIC protocol aims at resolving this issue since it is based on UDP and has built-in in support for connection migration (i.e. switching from WiFi to mobile network).
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 ;)
That is to say, if I have a server listening on 127.0.0.1, and a TCP connection comes in, how can I determine the process id of the client?
Also if there isn't an API for this, where would I be able to extract the information from in a more hackish manner?
(The purpose of this is to modify a local HTTP proxy server to accept or deny requests based on the requesting process.)
Edit: palacsint's answer below led me to find this answer to a similar question which is just what's needed
netstat -a -o
prints it. I suppose they are on the same machine becase you are listening on 127.0.0.1.
The only way to do this is if the connecting process sends some sort of custom headers which contains identifier. This is due to the fact that the networking layer is completely separated from the application layer (hint: OSI MODEL. This way it is possible to write lower layers software without caring what happens above as long as the messages exchanged (read: networking packets) follow a pre-determined format (read: use the same protocol).
In a recent series of question I have asked alot about UDP, boost::asio and c++ in general.
My latest question, which doesn't seem to have an answer here at Stackoverflow, is this:
In a client/server application, it is quite okay to require that the server open a port in any firewall, so that messages are allowed in. However, doing the same for clients is definately not a great user experience.
TCP-connections typically achieve this due to the fact that most routers support stateful packet inspection, allowing response packets through if the original request originated from the local host.
It is not quite clear to me how this would work with UDP, since UDP is stateless, and there is no such thing as "response packets" (to my knowledge). How should I account for this in my client application?
Thanks for any answers!
UDP itself is stateless, but the firewall typically is not. The convention on UDP is that if a request goes out from client:port_A to server:port_B, then the response will come back from server:port_B to client:port_A.
The firewall can take advantage of this. If it sees a UDP request go out from the client, it adds an entry to its state table that lets it recognise the response(s), to allow them in. Because UDP is stateless and has no indication of connection termination, the firewall will typically implement a timeout - if no traffic occurs between that UDP address pair for a certain amount of time, the association in the firewall's state table is removed.
So - to take advantage of this in your client application, simply ensure that your server sends responses back from the same port that it uses to receive the requests.