I have a question regarding the TCPStream package in Rust. I want to read data from a server. The problem is that it is not guaranteed that the data is sent in one TCP package.
And here comes my question:
Is the read message capable of reading more than one package, or do I have to call it more than one? Is there any "best practice"?
From the user space TCP packets are not visible and their boundaries don't matter. Instead user space reads only a byte stream and writes only to a byte stream. Packetizing is done at a lower level in a way to be optimal for latency and bandwidth. It might well happen that multiple write from user space end up in the same packet and it might also happen that a single write will result in multiple packets. And the same is true with read: it might get part of a packet, it might get the payload taken from multiple consecutive packets ...
Any packet boundaries from the underlying transport are no longer visible from user space. Thus protocols using TCP must implement their own message semantic on top of the byte stream.
All of this is not specific to Rust, but applies to other programming languages too.
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I have two apps sending tcp packages, both written in python 2. When client sends tcp packets to server too fast, the packets get concatenated. Is there a way to make python recover only last sent package from socket? I will be sending files with it, so I cannot just use some character as packet terminator, because I don't know the content of the file.
TCP uses packets for transmission, but it is not exposed to the application. Instead, the TCP layer may decide how to break the data into packets, even fragments, and how to deliver them. Often, this happens because of the unterlying network topology.
From an application point of view, you should consider a TCP connection as a stream of octets, i.e. your data unit is the byte, not a packet.
If you want to transmit "packets", use a datagram-oriented protocol such as UDP (but beware, there are size limits for such packets, and with UDP you need to take care of retransmissions yourself), or wrap them manually. For example, you could always send the packet length first, then the payload, over TCP. On the other side, read the size first, then you know how many bytes need to follow (beware, you may need to read more than once to get everything, because of fragmentation). Here, TCP will take care of in-order delivery and retransmission, so this is easier.
TCP is a streaming protocol, which doesn't expose individual packets. While reading from stream and getting packets might work in some configurations, it will break with even minor changes to operating system or networking hardware involved.
To resolve the issue, use a higher-level protocol to mark file boundaries. For example, you can prefix the file with its length in octets (bytes). Or, you can switch to a protocol that already handles this kind of stuff, like http.
First you need to know if the packet is combined before it is sent or after. Use wireshark to check it the sender is sending one packet or two. If it is sending one, then your fix is to call flush() after each write. I do not know the answer if the receiver is combining packets after receiving them.
You could change what you are sending. You could send bytes sent, followed by the bytes. Then the other side would know how many bytes to read.
Normally, TCP_NODELAY prevents that. But there are very few situations where you need to switch that on. One of the few valid ones are telnet style applications.
What you need is a protocol on top of the tcp connection. Think of the TCP connection as a pipe. You put things in one end of the pipe and get them out of the other. You cannot just send a file through this without both ends being coordinated. You have recognised you don't know how big it is and where it ends. This is your problem. Protocols take care of this. You don't have a protocol and so what you're writing is never going to be robust.
You say you don't know the length. Get the length of the file and transmit that in a header, followed by the number of bytes.
For example, if the header is a 64bits which is the length, then when you receive your header at the server end, you read the 64bit number as the length and then keep reading until the end of the file which should be the length.
Of course, this is extremely simplistic but that's the basics of it.
In fact, you don't have to design your own protocol. You could go to the internet and use an existing protocol. Such as HTTP.
I am wondering, if the speed of copy processes between user and kernel space, and in general within the whole tcp send/receive process, is dependent on the type of file (.txt, .mp4).
I mean not the file size, but the "structure" of the bytes or anything. I searched for quite a while but did not find anything related. Are there helpful phrases or terms I could have a look for ?
Thanks in advance!
TCP has no idea of the application level structure of a file and thus cannot make any decisions based on this. All what it cares about is a byte stream.
But it is not uncommon that how the application interacts with the kernel depends on the specific protocol and that this can have noticeable effects on the performance: for example one application might just write 1000 bytes at once to the kernel while another writes a 500 byte HTTP header followed by a 500 byte HTTP body. These might in both cases be the exact same bytes but more context switches are involved in the second case due to more syscalls and depending on the socket options this might also result in two TCP packets instead of one, where each TCP packet has a noticeable overhead in bytes and in processing time.
I am a newbie to networks and in particular TCP (I have been fooling a bit with UDP, but that's it).
I am developing a simple protocol based on exchanging messages between two endpoints. Those messages need to be certified, so I implemented a cryptographic layer that takes care of that. However, while UDP has a sound definition of a packet that constitutes the minimum unit that can get transferred at a time, the TCP protocol (as far as my understanding goes) is completely stream oriented.
Now, this puzzles me a bit. When exchanging messages, how can I tell where one starts and the other one ends? In principle, I can obviously communicate fixed length messages or first communicate the size of each message in some header. However, this can be subject to attacks: while of course it is going to be impossible to distort or determine the content of the communication, the above technique would make it easy to completely disrupt my communication just by adding a single byte in the middle.
Say that I need to transfer a message 1234567 bytes long. First of all, I communicate 4 bytes with an integer representing the size of the message. Okay. Then I start sending out the actual message. That message gets split in several packets, which get separately received. Now, an attacker just sends in an additional packet, faking it as if it was part of the conversation. It can just be one byte long: this completely destroys any synchronization mechanism I have implemented! The message has a spurious byte in the middle, and it doesn't successfully get decoded. Not only that, the last byte of the first message disrupts the alignment of the second message and so on: the connection is destroyed, and with a simple, simple attack! How likely and feasible is this attack anyway?
So I am wondering: what is the maximum data unit that can be transferred at once? I understand that to a call to send doesn't correspond a call to receive: the message can be split in different chunks. How can I group the packets together in some way so that I know that they get packed together? Is there a way to define an higher level message that gets reconstructed and aligned all together and triggers a single call to a receive-like function? If not, what other solutions can I find to keep my communication re-alignable even in presence of an attacker?
Basically it is difficult to control the way the OS divides the stream into TCP packets (The RFC defining TCP protocol states that TCP stack should allow the clients to force it to send buffered data by using push function, but it does not define how many packets this should generate. After all the attacker can modify any of them).
And these TCP packets can get divided even more into IP fragments during their way through the network (which can be opted-out by a 'Do not fragment' IP flag -- but this flag can cause that your packets are not delivered at all).
I think that your problem is not about introducing packets into a stream protocol, but about securing it.
IPSec could be very beneficial in your scenario, as it operates on the network layer.
It provides integrity for every packet sent, so any modification on-the-wire gets detected and the invalid packets are dropped. In case of TCP the dropped packets get re-transmitted automatically.
(Almost) everything is done automatically by the OS -- so yo do not need to worry about it (and make mistakes doing so).
The confidentiality can be assured as well (with the same advantage of not re-inventing the wheel).
IPSec should provide you a reliable transport protocol ontop of which you can use whatever framing format you like.
Another alternative is using SSL/TLS on top of TCP session which is less robust (as it does close the whole connection on integrity error).
Now, an attacker just sends in an additional packet, faking it as if it was part of the conversation. It can just be one byte long: this completely destroys any synchronization mechanism I have implemented!
Thwarting such an injection problem is dealt with by securing the stream. Create an encrypted stream and send your packets through that.
Of course the encrypted stream itself then has this problem; its messages can be corrupted. But those messages have secure integrity checks. The problem is detected, and the connection can be torn down and re-established to resynchronize it.
Also, some fixed-length synchronizing/framing bit sequence can be used between messages: some specific bit pattern. It doesn't matter if that pattern occurs inside messages by accident, because we only ever specifically look for that pattern when things go wrong (a corrupt message is received), otherwise we skip that sequence. If a corrupt message is received, we then receive bytes until we see the synchronizing pattern, and assume that whatever follows it is the start of a message (length followed by payload). If that fails, we repeat the process. When we receive a correct message, we reply to the peer, which will re-transmit anything we didn't get.
How likely and feasible is this attack anyway?
TCP connections are identified by four items: the source and destination IP, and source and destination port number. The attacker has to fake a packet which matches your stream in these four identifiers, and sneak that packet past all the routers and firewalls between that attacker and the receiving machine. The attacker also has to be in the right ballpark with regard to the TCP sequence number.
Basically, this is next to impossible for an attacker C to perpetrate against endpoints A and B which are both distant from C on the network. The fake source IP will be rejected long before C is able to reach its destination. It's more plausible as an inside job (which includes malware): C is close to A and B.
I know that it is possible that multiple packets would be stacked to the buffer to be read from and that a long packet might require a loop of multiple send attempts to be fully sent. But I have a question about packaging in these cases:
If I call recv (or any alternative (low-level) function) when there are multiple packets awaiting to be read, would it return them all stacked into my buffer or only one of them (or part of the first one if my buffer is insufficient)?
If I send a long packet which requires multiple iterations to be sent fully, does it count as a single packet or multiple packets? It's basically a question whether it marks that the package sent is not full?
These questions came to my mind when I thought about web sockets packaging. Special characters are used to mark the beginning and end of a packet which sorta leads to a conclusion that it's not possible to separate multiple packages.
P.S. All the questions are about TCP/IP but you are welcomed to share information (answers) about UDP as well.
TCP sockets are stream based. The order is guaranteed but the number of bytes you receive with each recv/read could be any chunk of the pending bytes from the sender. You can layer a message based transport on top of TCP by adding framing information to indicate the way that the payload should be chunked into messages. This is what WebSockets does. Each WebSocket message/frame starts with at least 2 bytes of header information which contains the length of the payload to follow. This allows the receiver to wait for and re-assemble complete messages.
For example, libraries/interfaces that implement the standard Websocket API or a similar API (such as a browser), the onmessage event will fire once for each message received and the data attribute of the event will contain the entire message.
Note that in the older Hixie version of WebSockets, each frame was started with '\x00' and terminated with '\xff'. The current standardized IETF 6455 (HyBi) version of the protocol uses the header information that contains the length which allows much easier processing of the frames (but note that both the old and new are still message based and have basically the same API).
TCP connection provides for stream of bytes, so treat it as such. No application message boundaries are preserved - one send can correspond to multiple receives and the other way around. You need loops on both sides.
UDP, on the other hand, is datagram (i.e. message) based. Here one read will always dequeue single datagram (unless you mess with low-level flags on the socket). Event if your application buffer is smaller then the pending datagram and you read only a part of it, the rest of it is lost. The way around it is to limit the size of datagrams you send to something bellow the normal MTU of 1500 (less IP and UDP headers, so actually 1472).
What's the difference between sockets (stream) vs sockets (datagrams)? Why use one over the other?
A long time ago I read a great analogy for explaining the difference between the two. I don't remember where I read it so unfortunately I can't credit the author for the idea, but I've also added a lot of my own knowledge to the core analogy anyway. So here goes:
A stream socket is like a phone call -- one side places the call, the other answers, you say hello to each other (SYN/ACK in TCP), and then you exchange information. Once you are done, you say goodbye (FIN/ACK in TCP). If one side doesn't hear a goodbye, they will usually call the other back since this is an unexpected event; usually the client will reconnect to the server. There is a guarantee that data will not arrive in a different order than you sent it, and there is a reasonable guarantee that data will not be damaged.
A datagram socket is like passing a note in class. Consider the case where you are not directly next to the person you are passing the note to; the note will travel from person to person. It may not reach its destination, and it may be modified by the time it gets there. If you pass two notes to the same person, they may arrive in an order you didn't intend, since the route the notes take through the classroom may not be the same, one person might not pass a note as fast as another, etc.
So you use a stream socket when having information in order and intact is important. File transfer protocols are a good example here. You don't want to download some file with its contents randomly shuffled around and damaged!
You'd use a datagram socket when order is less important than timely delivery (think VoIP or game protocols), when you don't want the higher overhead of a stream (this is why DNS is primarily a datagram protocol, so that servers can respond to many, many requests at once very quickly), or when you don't care too much if the data ever reaches its destination.
To expand on the VoIP/game case, such protocols include their own data-ordering mechanism. But if one packet is damaged or lost, you don't want to wait on the stream protocol (usually TCP) to issue a re-send request -- you need to recover quickly. TCP can take up to some number of minutes to recover, and for realtime protocols like gaming or VoIP even three seconds may be unacceptable! Using a datagram protocol like UDP allows the software to recover from such an event extremely quickly, by simply ignoring the lost data or re-requesting it sooner than TCP would.
VoIP is a good candidate for simply ignoring the lost data -- one party would just hear a short gap, similar to what happens when talking to someone on a cell phone when they have poor reception. Gaming protocols are often a little more complex, but the actions taken will usually be to either ignore the missing data (if subsequently-received data supercedes the data that was lost), re-request the missing data, or request a complete state update to ensure that the client's state is in sync with the server's.
Stream Socket:
Dedicated & end-to-end channel between server and client.
Use TCP protocol for data transmission.
Reliable and Lossless.
Data sent/received in the similar order.
Long time for recovering lost/mistaken data
Datagram Socket:
Not dedicated & end-to-end channel between server and client.
Use UDP for data transmission.
Not 100% reliable and may lose data.
Data sent/received order might not be the same.
Don't care or rapid recovering lost/mistaken data.
If it is the network programming I think starting from sockets would be a good start.
socket = ip + port
there are three types of sockets
stream (TCP, order and delivery guaranteed,no duplication,no length or char boundaries for data,connection-oriented,reliable, concurrency)
datagram(UDP,packet-based, connectionless, datagram size limit, data can be lost or duplicated, order not guaranteed,not reliable)
raw (direct access to lower layer protocols IP,ICMP)
I do not see any strict rule for transport protocol type as to what socket has to use what transport protocol and reliability should not be mistaken because UDP is realiable in case both ends are active.
Reliability refers to more like reliability of delivery since there are sequence number checks by using TCP as transport protocol which do not exist in UDP.It is better using network protocol analyzer like wireshark tcpdump etc to see what your software is exactly doing; kind of verification or merging theory on the paper with your work in action.