I am using Winsock, and I have a need to issue a TCP connect repeatedly to a third-party server. These applications will stay up potentially for days at a time. I am the only client connecting to the server. The time between connects is on the order of seconds, and the connection stays up only long enough to send a single message of a few bytes. I am currently seeing that the connects start to fail (WSAECONNREFUSED) after a few hours. Is there anything I must do (e.g. socket options, etc.) to ensure these frequent repeated connects will succeed for an indefinite amount of time? Thanks!
When doing a lot of transaction based connections and having issues with TCP's TIME_WAIT state duration (which last 2MSL = 120 seconds) leading to no more connections available for a client host toward a special server host, you should consider UDP and managing yourself the re-sending of lost requests.
I know that sounds odd. But standard services like DNS are required to use UDP to handle a ton of transactions (request then a single answer in one UDP segment) in order to avoid issues you are experimenting yourself. Web browsers send a request using UDP to the DNS. Re-request is done using UDP after a short time, no longer than a few milliseconds I guess. Sometimes the resolved name is too long and does not fit in the UDP paquet. As a consequence the DNS server send a UDP reply with a dedicated flag raised, in order to ask the client to use TCP this time.
Moreover you may consider also the T/TCP extension (Transactional TCP) of TCP, if available on your Windows platform. It provides TCP reliability with shorter TIME_WAIT state, as nearly no costs in the modifications of your client code. As far as I know it may work even though the server does not handle that extension. As a side note it is currently not used on the internet as it is know to have some flaw...
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I think it relates just to the TCP layer, but I describe my setup in the following paragraph:
On google compute engine I set up a http and websocket server (python, geventwebsocket+gevent.WSGIServer). At home I have my computer (esp8266) that connects to it using websockets.
I use websockets because I need bidirectional communication (a couple of messages a day, it goes like this: a message from server, a response from client.) The connection itself is initiated by the client, as it's behind a NAT.
The problem is that a couple of seconds from the last packet exchange, the messages from server don't arrive to the client. However, the client can send packets to the server even minutes after (and possibly much longer). And interestingly then, the probably retransmitted packets from server finally arrive.
I examined the packets are indeed sent from server with wireshark (and retrasmitted, if not ack'ed) and log every network communication on the client, so the problem probably isn't the application software. I get no exceptions in the applications. The connections are open.
I tested the time server can sent packets after the connection initiation/last delivered packet generally and it's between 6 and 20 seconds, varying between tests. In the test server sends out packets with a set, fixed, delay between them.
In a test (couple of packets) with the single set delay usually either all packets arrive, or none (yeah if one doesn't arrive, the next won't).
I suspect that might be because of the NAT. But then the one solution I see would be to periodically (every 6 seconds or less) send out keep alive packets (Pings and Pongs in websocket, or the TCP's keepalive) from the client. But that doesn't seem elegant, as there should be only a few data messages in a day.
And the similar thing happens when ssh'ing from my desktop to the server: after a couple seconds of inactivity at my and server side, the server stops sending anything (tested e.g. with watch -n20 date. Sometimes it just freezes and doesn't update until I press a key = send a packet from client. But the update is not instant in case of the ssh, it takes a couple of seconds after the keypress to see new stuff. Edit: of course that must be due to the retransmission timer algorithm)
So I studied what is the purpose of TCP keep-alive packets etc. and the thing is that routers and NAT's forget the connections or mappings or whatever in some time/keep only the newest. (So I guess in the case of client->server the mappings just recreate as the destination ip is public and is the actual server. And in the opposite direction it is not possible, so it doesn't work.)
But didn't think it can be as bad as in 6 seconds. The websockets almost reduce to polling (although with a possibly smaller lag).
It seems that the router's NAT mechanism may cause the problem. Maybe you can usee some little tools like NAT-PMP or Upnp to open a port and mapping to your local client. This will last long enough for you to do bidirectional communication.
I'm creating an IoT Device + Server system using .NET Micro Framework and ASP.NET WebAPI (Probably in Azure).
The IoT device needs to be able to frequently update the server with stats whilst also being able to receive occasional incoming commands from the server that would change its behaviour. In this sense, the device needs to act as both client and server itself.
My concern is getting the best balance between the security of the device and the load on the server. Furthermore, there must be a relatively low amount of latency between the server needing to issue a command and the device carrying it out; of the order of a few seconds.
As I see it my options are:
Upon connection to the internet, the device establishes a persistent TCP connection to the server which is then used for both polling and receiving commands.
The device listens on a port (e.g. HttpListener) for incoming commands whilst updating the server via frequent HTTP requests.
The device only ever polls the server with HTTP requests. The server uses the response to give the device commands.
The 2nd option seems to be the least secure as the device would have open incoming ports. The 1st option looks the most difficult to reliably implement as it would require low level socket programming. The 3rd option seems easy and secure but due to the latency requirements the device would need to poll every few seconds. This impacts the scalability of the system.
So at what frequency does HTTP polling create more overhead than just constantly keeping a TCP connection open? 5s? 3s? 1s? Or am I overstating the overhead of keeping a TCP connection open in ASP.NET? Or is there a completely different way that this can be implemented?
Thanks.
So at what frequency does HTTP polling create more overhead than just constantly keeping a TCP connection open? 5s? 3s? 1s?
There is nothing to do to keep a TCP connection open. The only thing you might need to do is to use TCP keep-alive (which have nothing to do with HTTP keep-alive!) in case you want to keep the connection idle (i.e no data to send) for a long time.
with HTTP your overhead already starts with the first request, since your data need to be encapsulated into a HTTP message. This overhead can be comparable small if the message is large or it can easily be much larger than the message itself for small messages. Also, HTTP server close the TCP connection after some idle time so you might need to re-establish the TCP connection for the next data exchange which is again overhead and latency.
HTTP has the advantage to pass through most firewalls and proxies, while plain TCP does not. You also get encryption kind of free with HTTPS, i.e. there are established standards for direct encrypted connection and for tunneling through a proxy.
WebSockets is something in between: you do a HTTP request and then upgrade HTTP to WebSocket. The initial overhead is thus as large as for HTTP but for the next messages the overhead is not that much higher than TCP. And you can do also WebSockets with HTTPS (i.e. wss:// instead of ws://). It passes through most simple firewalls and proxies, but more deeper inspection firewalls might still have trouble with it.
Setting up a TCP listener will be a problem if you have your IoT device behind some NAT router, i.e. the usual setup inside private or SoHo networks. To reach the device one would need to open a tunnel at the router from outside into the network, either by administrating the router by hand or with UPnP (which is often switched off for security reasons). So you would introduce too much problems for the average user.
Which means that the thing which the fewest problems for the customer is probably HTTP polling. But this is also the one with the highest overhead. Still mostly compatible are WebSockets which have less overhead and more problems but even less overhead can be reached with simple TCP to the server. TCP listener instead would cause too much trouble.
As for resources on the server side: each HTTP polling request might use new TCP connection but you can also reuse an existing one. In this case you could decide between more overhead and latency one the client side (new TCP connection for each request) which needs few resources on the server side and less overhead and latency on the client side which needs more resources on the server side (multiple HTTP requests per TCP connection). With WebSockets and plain TCP connection you always need more server side resources, unless your client will automatically re-establish the connection on loss of connectivity.
These days you should use a IOT Specific communication protocol over TLS 2.0 for secure light weight connections. For example AWS uses MQTT http://mqtt.org/ and Azure uses AMQP https://www.amqp.org/
The idea is you get a broker you can connect to securely then you use a messaging protocol with a topic to route messages to the proper devices. Also IBM has been using MQTT for a long time and routers now typically come with port 8883 open which is MQTT over TLS.
Good Luck!
Simply use SignalR to connect client and server. It provides you minimal latency without polling. The API is very simple to use.
Physically, this runs over WebSockets which are scalable to a large number of concurrent connections. If you don't have a need for more than 100k per Windows server this would not be a concern.
I'm not sure if this is the correct place to ask, so forgive me if it isn't.
I'm writing computer monitoring software that needs to connect to a server. The server may send out relatively urgent messages, such as sound or cancel an alarm, and the client may send out data about the computer, such as screenshots. The data that the client sends isn't too critical on timing, but shouldn't be more than a two minutes late.
It is essential to the software that portforwarding need not be set up, and it is assumed that the internet connection will be done through a wireless router that has NAT almost all the time.
My idea is to have a TCP connection initiated from the client, and use that to transfer data. Ideally, I would have no data being sent when it is not needed, but I believe this to be impossible. Would sending the equivalent of a ping every now and again keep the connection alive, and what sort of bandwidth would it use if this program was running all the time on the computer? In addition, would it be possible to reduce the header size for these keep-alives?
Before I start designing the communication and programming, is this plan for connection flawed? Are there better alternatives?
Thanks!
1) You do not need to send 'ping' data to keep the connection alive, the TCP stack does this automatically; one reason for sending 'ping' data would be to detect a connection close on the client side - typically you only find out something has gone wrong when you try and read/write from the socket. There may be a way to change various time-outs so you can detect this condition faster.
2) In general while TCP provides a stream-oriented error free channel, it makes no guarantees about timeliness, if you are using it on the internet it is even more unpredictable.
3) For applications such as this (I hope you are making it for ethical purposes) - I would tend to use TCP, since you don't want a situation where the client receives a packet to raise an alarm but misses that one that turns it off again.
Whats the best practice for scalable servers which need to maintain a list of active users?
Should I open a persistent TCP Connection for each client on which the server sends update messages?
This could lead in many open connection and propably no traffic for many seconds. Is this a problem in TCP?
Or would it be better to let the Client poll updates periodically (with a new tcp connection each)?
How do Chat Servers or large Online Games handle this?
Personally I'd go for a single persistent TCP connection per client to avoid a) the additional work in creating and destroying connections and the additional latency involved in all the TCP packets involved and b) to avoid creating lots of sockets in TIME_WAIT on either the clients or the server. There's simply no good reason to create and destroy the connections.
Depending on your platform there may be various tricks to deal with the various platform specific problems that you might get when you have lots of connections open, and by lots I mean 10s of thousands. For example, on Windows, using overlapped I/O and I/O completion ports would be a good design for lots of connections and if your connections are generally idle most of the time then you might find that using the 'zero byte read' trick would allow you to handle more connections on lesser hardware; but it's something you can add once you know you have a problem due to the amount of buffer space that you have waiting for reads which only complete infrequently.
I wouldn't have the clients polling the server. It's inefficient. Have the server publish data to the clients as and when there is data available. This would allow the server to control the workload somewhat by letting it decide how often to send the data to the clients - it could either send every time new data became available for a client or send after it had batched up some data and waited a short while, etc. If the server is pushing the data then the server (the weak point, the place that might get overwhelmed by client demand) has more control over the work that it will need to do.
If you have each client polling then a) you're generating more network noise as each client sends a message to ask the server if it has anything that it should send it and b) you're generating more work for the server as it needs to respond to the polls. The server knows when there's data for the client, let it be responsible for telling the clients.
We're trying to tune an application that accepts messages via TCP and also uses TCP for some of its internal messaging. While load testing, we noticed that response time degrades significantly (and then stops altogether) as more simultaneous requests are made to the system. During this time, we see a lot of TCP connections in TIME_WAIT status and someone suggested lowering the TIME_WAIT environment variable from it's default 60 seconds to 30.
From what I understand, the TIME_WAIT setting essentially sets the time a TCP resource is made available to the system again after the connection is closed.
I'm not a "network guy" and know very little about these things. I need a lot of what's in that linked post, but "dumbed down" a little.
I think I understand why the TIME_WAIT value can't be set to 0, but can it safely be set to 5? What about 10? What determines a "safe" setting for this value?
Why is the default for this value 60? I'm guessing that people a lot smarter than me had good reason for selecting this as a reasonable default.
What else should I know about the potential risks and benefits of overriding this value?
A TCP connection is specified by the tuple (source IP, source port, destination IP, destination port).
The reason why there is a TIME_WAIT state following session shutdown is because there may still be live packets out in the network on their way to you (or from you which may solicit a response of some sort). If you were to re-create that same tuple and one of those packets showed up, it would be treated as a valid packet for your connection (and probably cause an error due to sequencing).
So the TIME_WAIT time is generally set to double the packets maximum age. This value is the maximum age your packets will be allowed to get to before the network discards them.
That guarantees that, before you're allowed to create a connection with the same tuple, all the packets belonging to previous incarnations of that tuple will be dead.
That generally dictates the minimum value you should use. The maximum packet age is dictated by network properties, an example being that satellite lifetimes are higher than LAN lifetimes since the packets have much further to go.
Usually, only the endpoint that issues an 'active close' should go into TIME_WAIT state. So, if possible, have your clients issue the active close which will leave the TIME_WAIT on the client and NOT on the server.
See here: http://www.serverframework.com/asynchronousevents/2011/01/time-wait-and-its-design-implications-for-protocols-and-scalable-servers.html and http://www.isi.edu/touch/pubs/infocomm99/infocomm99-web/ for details (the later also explains why it's not always possible due to protocol design that doesn't take TIME_WAIT into consideration).
Pax is correct about the reasons for TIME_WAIT, and why you should be careful about lowering the default setting.
A better solution is to vary the port numbers used for the originating end of your sockets. Once you do this, you won't really care about time wait for individual sockets.
For listening sockets, you can use SO_REUSEADDR to allow the listening socket to bind despite the TIME_WAIT sockets sitting around.
In Windows, you can change it through the registry:
; Set the TIME_WAIT delay to 30 seconds (0x1E)
[HKEY_LOCAL_MACHINE\SYSTEM\CurrentControlSet\Services\TCPIP\Parameters]
"TcpTimedWaitDelay"=dword:0000001E
setting the tcp_reuse is more useful than changing time_wait, as long as you have the parameter (kernels 3.2 and above, unfortunately that disqualifies all versions of RHEL and XenServer).
Dropping the value, particularly for VPN connected users, can result in constant recreation of proxy tunnels on the outbound connection. With the default Netscaler (XenServer) config, which is lower than the default Linux config, Chrome will sometimes have to recreate the proxy tunnel up to a dozen times to retrieve one web page. Applications that don't retry, such as Maven and Eclipse P2, simply fail.
The original motive for the parameter (avoid duplication) was made redundant by a TCP RFC that specifies timestamp inclusion on all TCP requests.
I have been load testing a server application (on linux) by using a test program with 20 threads.
In 959,000 connect / close cycles I had 44,000 failed connections and many thousands of sockets in TIME_WAIT.
I set SO_LINGER to 0 before the close call and in subsequent runs of the test program had no connect failures and less than 20 sockets in TIME_WAIT.
TIME_WAIT might not be the culprit.
int listen(int sockfd, int backlog);
According to Unix Network Programming Volume1, backlog is defined to be the sum of completed connection queue and incomplete connection queue.
Let's say the backlog is 5. If you have 3 completed connections (ESTABLISHED state), and 2 incomplete connections (SYN_RCVD state), and there is another connect request with SYN. The TCP stack just ignores the SYN packet, knowing it'll be retransmitted some other time. This might be causing the degradation.
At least that's what I've been reading. ;)