I am using Wireshark to capture TCP communication packets in my experiment (using Mininet Simulation). The experiment involves a client (10.0.0.2) who requests HTTP page from the server (10.0.0.1). The following screenshot shows the capturing number 75 when the client sends HTTP GET and capturing number 89 when the server responds. Between them, I don't know what these packets are. Are they important in terms of TCP communication?
Thank you
depends on what you are analyzing. If you are concerned with the HTTP conversation then right click on one of the http rows and click- follow TCP stream. This lets you see entire http session. If you are having issues with the underlying TCP connection then those in-between rows become important.
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SSEs are advertised as a unidirectional communication tool to be used from server to client. I have a requirement to broadcast data to all clients and so i was wondering how SSEs behave on a low level. I cannot seem to find any low level information about SSEs online.
Primarily i would like to know if, after sending the data, does the server wait for a response from the client to confirm it has received the data before finishing the "send". That would mean that doing a broadcast using a for loop would be quiet dangerous and slow in which case websockets might be the better options.
Perhaps the implementation depends entirely on the language and framework? Is it not standardized?
Broadcast usually uses UDP which does not wait for a response. - - Broadcasting ip:port by socket server
.. says
UDP Packet: First four bytes as a magic number, next four bytes an IPv4 address (and you might want to add other things like a server name).
The magic number is just in case there is a collision with another application using the same port. Check both the length of the packet and the magic number.
Server would broadcast the packet at something like 30 second time intervals. (Alternatively you could have the server send a response only when a client sends a request via broadcast.)
So the client app would have to send a request back to the server app.
Different protocols would get different responses according the the underlying technology. eg HTTP uses responses extnsivly.
SSE and WebSockets are both over TCP, so there could be a wait before the socket could be used to send further data.
However, each client is a dedicated socket. So server-side you would be using threads or async coding (depending on the server-side language and its conventions). So looping through all the sockets to send a message to each client would be fine and quick.
I'm studying Socket Programming HOWTO and the author at some point says that
A protocol like HTTP uses a socket for only one transfer.
Is it because of the design of the HTTP protocol itself? Or is it because it is based on TCP, so all protocols based on it (e.g. UDP) must use one socket for only one transfer?
This statement is taken out of context. The context is to point out that TCP is not a message based protocol but an unstructured byte stream. And to have a message semantic one needs to have some way to determine where a message ends.
It then takes HTTP as an example where a message might simply end with a connection close and points out the limitations - namely only a single message per connection per direction. Then it goes on to describe how protocols can be designed without this limitation, i.e. having multiple messages per connection.
HTTP still can be used like this, i.e. have a single request and end with connection close. This is the design of HTTP version 0.9, but can still be done with HTTP/1. But with HTTP/1 it can also be used for multiple messages, one after the other. And with HTTP/2 it can do multiple messages in parallel, multiplexed over a single TCP connection. And HTTP/3 does not even use TCP anymore.
Do all protocols based on TCP use one socket per transfer?
Protocols are not limited to one connection ("socket") per message ("transfer"). Depending on the design of the protocol multiple messages can be send one after the other by having some pre-known message size or a clear message delimiter. Some protocols might send multiple messages in parallel by implementing a multiplexing layer on top of TCP. Some protocols might even use multiple TCP connections in parallel to deliver a single message, i.e. distributing the message over multiple connections.
That statement was probably written in 1996 or earlier. Since 1997, HTTP supports persistent connections, reusing the same TCP connection and the same socket for multiple queries.
I captured packets using Wireshark while my computer is connecting to random web site for my assignment using TCP. Normally, there should be 3 initial packets (SYN, SYN-ACK, ACK) for 3-way handshake before the data transfer starts.
In my case, there are 9 connections, 3 different 3-way handshake; each was done with a different port of my computer. After each of them are accepted, the data transfer continues with only one of the ports.
I couldn't understand the reason behind this. Thanks!
It is common that browsers open multiple connections to a site. Reason for this is that with HTTP/1 a single TCP connection can only handle a single request-response at a time, i.e. multiple requests after each other and not in parallel. To handle multiple requests in parallel a browser needs to have multiple TCP connections open. Since connection setup takes some time browsers might open multiple connections to the server just to have some already established if new requests need to be send.
Client Send SYN with client isn.
Server reply SYN ACK with server's isn.
Client resend SYN if timeout?
When client send data, it can accumulated confirm the server's isn.
I try to search, but can't find the answer.
I know how the tcp is designed now, I just don't know why it's designed like this. Why can't use a two way handshake.
It cannot use a two way handshake by definition. TCP/IP is formalized as a standard for communication across networks (Internetworking). Specifically, RFC 793 requires that:
The "three-way handshake" is the procedure used to establish a
connection. This procedure normally is initiated by one TCP and
responded to by another TCP. The procedure also works if two TCP
simultaneously initiate the procedure. When simultaneous attempt
occurs, each TCP receives a "SYN" segment which carries no
acknowledgment after it has sent a "SYN". Of course, the arrival of
an old duplicate "SYN" segment can potentially make it appear, to the
recipient, that a simultaneous connection initiation is in progress.
Proper use of "reset" segments can disambiguate these cases.
If you think about how the protocol works, you do not actually have a single full duplex connection. Instead, you have two simplex connections, each going in one direction. This is why the proper response to a SYN is a SYN-ACK. The server is acknowledging the synchronization request by sending the originators sequence number as the acknowledgement number plus one; it is simultaneously attempting to open its own connection to the client by sending a SYN request to synchronize.
The proper answer to a SYN will always be an ACK, even if the connection fails.
Regarding your question about sending a retry, yes; the client will send a retry (typically up to three, but it could be as many as eight in practice... There's no limit defined) attempting to elicit a response of some kind (preferably a SYN-ACK, but possibly a RST-ACK).
We are trying to implement a proxy proof of concept but have encountered an interesting question: Since a single HTTP connection can, and indeed should, make multiple requests, and the HTTP transactions are sent via multiple packets due to TCP's magic, is it possible for a HTTP request to begin in the middle of a packet?
Bear in mind that this is not a theoretical question regarding possible optimization of the browser, but whether it actually happens in real life. It would be even better if someone could point me to a written reference on whether or not this is possible and if so how often it can occur.
Clarification update: We know that if we work in the HTTP layer alone we would not need to bother with this question, however we're trying to figure out if some advanced technique could be applied by working on the TCP layer first.
Assuming that you are talking about IP packets: Yes, it is possible that HTTP request starts middle of IP packet.
When you are using persistent HTTP connections, that is, using same TCP connection for several HTTP requests, it is fully possible that request boundary is middle of IP packet.
Also there is a TCP protocol between IP and HTTP. TCP contains also some headers so a IP packet may start with some TCP headers and rest of the packet consists of HTTP request.
HTTP request may also consist of several IP packets (in case of file uploads, transmission errors and following retransmissions etc).
However, I wonder why you are interested in packets if you are working at HTTP level. TCP should hide the IP packet details.
First of all, TCP is a stream based protocol and has no concept of packets. HTTP itself might have some kind of message or record delimiter, but TCP doesn't.
This page might be helpful: Structure of HTTP Transactions
From your question it sounds like you think that each read from a TCP socket is a "packet" of data. In reality, each read simply reads as many bytes as are in the buffer up to the maximum that you requested, without any concept of records or packets.
So for instance, lets say you read 2048 bytes from the socket, you could have the tail end of one transaction, followed by the beginning of a second response half way through the data you read, and only get the remainder of your second response on your next read from the socket.
If you're here in Jerusalem or near by maybe I could help you out.
Unless you are implementing your own TCP stack, you should not need to worry about the packets, but rather about the API that the TCP provides, in case of POSIX interfaces it would be the recv() or read(). So I treat the question then as "Can more than one HTTP requests come into a single read(), and can the HTTP request be split between multiple read() requests?" -- The answer to both would be "yes, it is possible".
An example of where this can happen is HTTP pipelining. This not frequent in real life (ironically, at least some of the browsers disable it by default because of "buggy proxies" :-) - but when it happens, can be a bit of a problem for the users to diagnose - especially if they have no access to the proxy.
One very notable place where it does happen by default apt-get in Debian-derived linux systems. Just install a Debian or Ubuntu server and try to use it through your proxy. You can do that by editing the /etc/apt/apt.conf.d/proxy file and placing the following there:
Acquire::http::Proxy "http://your.proxy.address:8080";
Depends of which abstraction layer of a packet you are talking about: there are many layers underneath HTTP.
HTTP --> TCP (byte stream) --> IP (packet) --> (possibly something else) Ethernet (frame) --> (possibly) some other transport
If you are talking about the IP layer, then yes the HTTP layer would start later on... Note that TCP presents a "byte stream interface" to its Client layer hence, no concept of packet here.
I think I understand where you are trying to go with this question.
If you don't use persistent HTTP connections, the HTTP GET request header is always the very first thing which is sent over the TCP connection, so we can be sure that the start of the HTTP GET request header does "not start in the middle of some TCP packet". But keep in mind that there may be one or more TCP packets without any user data, e.g. only a SYN, which may preceed the TCP packet with the start of the HTTP GET request header. And also keep in mind that the HTTP GET request header may not be contained in a single TCP packet.
If you do use persistent HTTP connections, the start of the HTTP GET request header for request number N+1 can start in the middle of a TCP packet, namely after the end of HTTP GET request body of request number N.
If you are asking these questions you are possibly "doing it wrong". As several other responders have already pointed out, in the vast majority of cases you should probably just be a TCP client and deal with a TCP stream of data and let the TCP code worry about the TCP packets. (Unless, of course, you are working on some special hardware which is looking at individual IP packets as they fly by and try to do some processing at the HTTP layer.)