Netty SSL mode strange behavior - ssl

I am trying to understand, why does Netty SSL mode work on strange way?
Also, the problem is following, when any SSL client(https browser, java client using ssl, also any ssl client application) connects to Netty server I get on beginning the full message, where I can recognize correctly the protocol used, but as long the channel stays connected, any following messages have strange structure, what is not happening same way with non-ssl mode.
As example on messageReceived method when the https browser connects to my server:
I have used PortUnificationServerHandler to switch protocols.. (without using nettys http handler, it is just example, because i use ssl mode for my own protocol too)
first message is ok, I get full header beginning with GET or POST
than I send response...
second message is only one byte long and contains "G" or "P" only.
third message is than the rest beginning either with ET or OST and the rest of http header and body..
here again follows my response...
fourth message is again one byte long and again contains only one byte..
fifth message again the rest... and on this way the game goes further..
here it is not important, which sub protocol is used, http or any else, after first message I get firstly one byte and on second message the rest of the request..
I wanted to build some art of proxy, get ssl data and send it unencoded on other listener, but when I do it directly without waiting for full data request, the target listener(http server as example) can not handle such data, if the target gets one byte as first only (even if the next message contains the rest), the channel gets immediately closed and request gets abandoned..
Ok, first though would be to do following, cache the first byte temporarily and wait for next message and than join those messages, and only than response, that works fine, but sometimes that is not correct approach, because the one byte is sometimes really the last message byte, and if i cache it and await wrongly next message, i can wait forever, because the https browser expects at this time some response and does not send any data more..
Now the question, is it possible to fix this problem with SSL? May be there are special settings having influence on this behavior?
I want fully joined message at once as is and not firstly first byte and than the rest..
Can you please confirm, that with newer Netty versions you have same behaving by using PortUnificationServerHandler (but without netty http handler, try some own handler.)
Is this behavior Ok so, I do not believe, it was projected so to work..

What you're experiencing is likely to be due to the countermeasures against the BEAST attack.
This isn't a problem. What seems to be the problem is that you're assuming that you're meant to read data in terms of messages/packets. This is not the case: TCP (and TLS/SSL) are meant to be used as streams of continuous data. You should keep reading data while data is available. Where to split incoming data where it's meaningful is guided by the application protocol. For HTTP, the indications are the blank line after the header and the Content-Length or chunked transfer encoding for the entity.
If you define your own protocol, you'll need a similar mechanism, whether you use plain HTTP or SSL/TLS. Assuming you don't need it only works by chance.

I had experienced this issue and found it was caused bu using JDK1.7. Moving back to JDK1.6 solved it. I did not have time to investigate further but have assumed for now that the SSLEngine implementation has changed in the JDK. I will investigate further when time permits.

Related

Handling Telnet negotiation

I'm trying to implement Telnet Client using C++ and QT as GUI.
I have no idea to handling the telnet negotiations.
Every telnet command is preceded by IAC, e.g.
IAC WILL SUPPRESS_GO_AHEAD
The following is how I handling the negotiation.
Search for IAC character in received buffer
According to the command and option, response to the request
My questions are described as follows:
It seems that the telnet server won't wait for a client response after a negotiation command is sent.
e.g. (send two or more commands without waiting for client reponse)
IAC WILL SUPPRESS_GO_AHEAD
IAC WILL ECHO
How should I handle such situation? Handle two requests or just the last one?
What the option values would be if I don't response the request? Are they set as default?
Why IAC character(255) won't be treated as data instead of command?
Yes, it is allowed to send out several negotiations for different options without synchronously waiting for a response after each of them.
Actually it's important for each side to try to continue (possibly after some timeout if you did decide to wait for a response) even if it didn't receive a reply, as there are legitimate situations according to the RFC when there shouldn't or mustn't be a reply and also the other side might just ignore the request for whatever reason and you don't have control over that.
You need to consider both negotiation requests the server sent, as they are both valid requests (you may choose to deny one or both, of course).
I suggest you handle both of them (whatever "handling" means in your case) as soon as you notice them, so as not to risk getting the server stuck if it decides to wait for your replies.
One possible way to go about it is presented by Daniel J. Bernstein in RFC 1143. It uses a finite state machine (FSM) and is quite robust against negotiation loops.
A compliant server (the same goes for a compliant client) defaults all negotiable options to WON'T and DON'T (i.e. disabled) at the start of the connection and doesn't consider them enabled until a request for DO or WILL was acknowledged by a WILL or DO reply, respectively.
Not all servers (or clients for that matter) behave properly, of course, but you cannot anticipate all ways a peer might misbehave, so just assume that all options are disabled until enabling them was requested and the reply was positive.
I'll assume here that what you're actually asking is how the server is going to send you a byte of 255 as data without you misinterpreting it as an IAC control sequence (and vice versa, how you should send a byte of 255 as data to the server without it misinterpreting it as a telnet command).
The answer is simply that instead of a single byte of 255, the server (and your client in the opposite direction) sends IAC followed by another byte of 255, so in effect doubling all values of 255 that are part of the data stream.
Upon receiving an IAC followed by 255 over the network, your client (and the server in the opposite direction) must replace that with a single data byte of 255 in the data stream it returns.
This is also covered in RFC 854.

About losing HTTP Requests

I have a server to which my client sends a HTTP GET request with some values. The server on its end simply stores these values to a database.
Now, I am observing that sometimes I do not observe these values in the database. One of the following could have happened:
The client never sent it
The server never received it
The server failed in writing to the database
My strongest doubt is that the reason is 2 - but I am unable to explain it completely. Since this is an HTTP request (which means there is TCP underneath) reliable delivery of the GET request should be guaranteed, right? Is it possible that even though I send a GET request to the server - it was never received by the server? If yes, what is TCP doing there?
Or, can I confidently assert that if the server is up and running and everything sent to the server is written to the database, then the absence of the details of the GET request in the database means the client never sent it?
Not sure if the details will help - but I am running a tomcat server and I am just sending a name-value pair through the get request.
There are a few things you seem to be missing. First of all, yes, if TCP finishes successfully, you pretty much have a guarantee that your message (i.e. the TCP payload) has reached the other side: TCP assures that it will take care of lost packages and the order in which packages arrive. However, this is not universially failproof, as there are still things beyond the powers of TCP (think of a physical disconnect by cutting through an ethernet cable). There is also no assertion regarding the syntactical correctness of the protocol "above." Any checks beyond delivering a bit-perfect copy is simply not TCP's concern.
So, there is a chance that the requests issued by your client are faulty or that they are indeed correct but not parsed correctly by your server. Former is striking me as more likely as latter one as Tomcat is a very mature piece of software. I think it would help tremendously if you would record and analyse some of your generated traffic through e.g. Wireshark.
You do not really mention what database you have in use. But there are some sacrificing acid-compliance in favour of increased write speeds. The nature of these databases brings it that you can never be really sure wether something actually got written to disk or is still residing in some buffer in memory. Should you happen to use such a db, this were another line of investigation.
Programmatically, I advise you take the following steps when dealing with HTTP traffic:
Has writing to the socket finishes without error?
Could a response be read from the socket?
Does the response carry a code in the 2xx range (indicating a successful operation)?
If any of these fail, you should really log something.
On a realated note, what you are doing there does not call for the GET method but for POST as you are changing application state. Consider it as a nice-to-have ;)
Without knowing the specifics, you can break it down into two parts. The HTTP request and the DB write. The client will receive a 200 OK response from the server when its GET request has been acknowledged. I've written code under Tomcat to connect to a MySQL DB using DAO. In the case of a failure an exception would be thrown and logged. Which ever method you're using, you'll want to figure out how failures are logged.

What exactly is the HTTP_EXPECT header for?

What exactly is the HTTP_EXPECT header for?
I have noticed it popping up in some but not all of my logs, and am curious what it means?
I never heard of this header before, but a little research was illuminating.
You can read the spec for HTTP_EXPECT right here. However, the spec language is perhaps a little dry and technical. The looser explanation is as follows:
Clients can send HTTP requests with large bodies. This would be a big waste of bandwidth if it turns out the server was going to end up rejecting the request for some reason (e.g. no user is logged in or similar). Enter the HTTP_EXPECT header. This header essentially lets the client make special demands of the server; "I the client expect that you will pre-approve this message, else… I won't even bother schlepping the big stuff over." On the server side, servers can usually choose to respond in one of two ways:
bounce back a 100 (please continue!) response, or
417 (nope, expectation failed) response.
This has to be done by the server before the big expensive request body follows down the stream pipe, so 100 is kind of an intermediate response before your final response (probably 200), and 417 is a final response that closes the pipe entirely so the client never ends up streaming the data.
Now, in theory the expectation header is extensible so it might be used for other purposes. The client could send an expect for something other than 100. But the above use case seems to be the primary (if not only) purpose of this header. See here for more info on the 100 response.
Of particular note to JS developers, I think it's worth pointing out that Node has 100-continue support, but as of today, doesn't yet support other Expect headers.

Why does the TLS heartbeat extension allow user supplied data?

The heartbeat protocol requires the other end to reply with the same data that was sent to it, to know that the other end is alive. Wouldn't sending a certain fixed message be simpler? Is it to prevent some kind of attack?
At least the size of the packet seems to be relevant, because according to RFC6520, 5.1 the heartbeat message will be used with DTLS (e.g. TLS over UDP) for PMTU discovery - in which cases it needs messages of different sizes. Apart from that it might be simply modelled after ICMP ping, where you can also specify the payload content for no reason.
Just like with ICMP Ping, the idea is to ensure you can match up a "pong" heartbeat response you received with whichever "ping" heartbeat request you made. Some packets may get lost or arrive out of order and if you send the requests fast enough and all the response contents are the same, there's no way to tell which of your requests were answered.
One might think, "WHO CARES? I just got a response; therefore, the other side is alive and well, ready to do my bidding :D!" But what if the response was actually for a heartbeat request 10 minutes ago (an extreme case, maybe due to the server being overloaded)? If you just sent another heartbeat request a few seconds ago and the expected responses are the same for all (a "fixed message"), then you would have no way to tell the difference.
A timely response is important in determining the health of the connection. From RFC6520 page 3:
... after a number of retransmissions without
receiving a corresponding HeartbeatResponse message having the
expected payload, the DTLS connection SHOULD be terminated.
By allowing the requester to specify the return payload (and assuming the requester always generates a unique payload), the requester can match up a heartbeat response to a particular heartbeat request made, and therefore be able to calculate the round-trip time, expiring the connection if appropriate.
This of course only makes much sense if you are using TLS over a non-reliable protocol like UDP instead of TCP.
So why allow the requester to specify the length of the payload? Couldn't it be inferred?
See this excellent answer: https://security.stackexchange.com/a/55608/44094
... seems to be part of an attempt at genericity and coherence. In the SSL/TLS standard, all messages follow regular encoding rules, using a specific presentation language. No part of the protocol "infers" length from the record length.
One gain of not inferring length from the outer structure is that it makes it much easier to include optional extensions afterwards. This was done with ClientHello messages, for instance.
In short, YES, it could've been, but for consistency with existing format and for future proofing, the size is spec'd out so that other data can follow the same message.

Twisted - succes (or failure) callback for LineReceiver sendLine

I'm still trying to master Twisted while in the midst of finishing an application that uses it.
My question is:
My application uses LineReceiver.sendLine to send messages from a Twisted TCP server.
I would like to know if the sendLine succeeded.
I gather that I need to somehow add a success (and error?) callback to sendLine but I don't know how to do this.
Thanks for any pointers / examples
You need to define "succeeded" in order to come up with an answer to this.
All sendLine does immediately (probably) is add some bytes to a send buffer. In some sense, as long as it doesn't raise an exception (eg, MemoryError because your line is too long or TypeError because your line was the number 3 instead of an actual line) it has succeeded.
That's not a very useful kind of success, though. Unfortunately, the useful kind of success is more like "the bytes were added to the send buffer, the send buffer was flushed to the socket, the peer received the bytes, and the receiving application acted on the data in a persistent way".
Nothing in LineReceiver can tell you that all those things happened. The standard solution is to add some kind of acknowledgement to your protocol: when the receiving application has acted on the data, it sends back some bytes that tell the original sender the message has been handled.
You won't get LineReceiver.sendLine to help you much here because all it really knows how to do is send some bytes in a particular format. You need a more complex protocol to handle acknowledgements.
Fortunately, Twisted comes with a few. twisted.protocols.amp is one: it offers remote method calls (complete with responses) as a basic feature. I find that AMP is suitable for a wide range of applications so it's often safe to recommend for new development. It largely supersedes the older twisted.spread (aka "PB") which also provides both remote method calls and remote object references (and is therefore more complex - in my experience, more complex than most applications need). There are also some options that are a bit more standard: for example, Twisted Web includes an HTTP implementation (HTTP, as you may know, is good at request/response style interaction).