I am trying to implement one tcp stack.
Following steps I have followed:
TCP open:
a. Client sent 'SYN' with a initial sequence no "C" and Ack no as "0" to server.
b. Server responded with SYN + ACK with seq no "S" and ack no "C+1".
c. Client send ACK+PSH with sequence no "C+1" and ack no "S+1".
TCP send and recv:
a. Client will send the request for data with ACK flag and the data
with seq no "C+1" and ack no "S+1"
b. The sender will send gragment with seq no "S+1+data" in segment and receiver will send ACK with ack no "S+1+data" will continue till the FIN or RST is transmitted.
In my case during data transmission I am not getting few packets but I can see them on wireshark.
Is there any mechanism to check for out of order data and packet loss using sequence number and ack number and to get the packet back
If you receive a segment with a sequence number higher than the next sequence expected, you should resend the last ACK. This tells the sender that some of the segments were lost, and it should immediately retransmit all the segments that are waiting for acknowledgement.
A better solution would be to implement the Selective Acknowledgement option in RFC 2018. This will allow you to acknowledge the segments received out of order, and the sender will only retransmit the ones that were missed.
Related
I have made a simple program in C.
Currenty I am trying to establish TCP handshake between server and client.
So when I recieve [SYN] flag then I respond with [SYN+ACK] flags (along with any sequence number, so I am choosing it as zero) and any ACK number
And in code when I receive in server ACK flag from client in TCP handshake's final packet I send ACK flag along with my own chosen sequence number (any number) and ACK numebr = one incremented of received SYC packet's sequence number because no data has been transmitted from server.
But I am not getting any ACK packet. my client is keep sending SYN packets and my server keep responding with Syn +ACK packet.
I like to know are sequence numbers and acknowledgement number during TCP handshake process are irreverent so I can focus on other fields which may be why my client ignoring my SYN responded SYC+ACK packet like IP and tcp checksum. Or is there anything wrong with my sequence and acknowlegement number handling first
In addition I am using TUN device in Linux that I created from server program
TCP sequence and acknowledgement numbers are absolutely NOT irrelevant during handshake (or any time in TCP connection). In fact, TCP handshake is there so that the server and the client can learn each other's sequence numbers. Your packets are ignored because ACKs do not match.
The ACK number in SYNACK packet is the sequence number in first SYN + 1. And the ACK number in last ACK packet is the seq. in SYNACK packet + 1.
In addition to the standard, linked by #Barmar, you can use this picture (source)
I'm writing udp server/client application in which server is sending data and client
is receiving. When packet is loss client should sent nack to server. I set the socket as
O_NONBLOCK so that I can notice if the client does not receive the packet
if (( bytes = recvfrom (....)) != -1 ) {
do something
}else{
send nack
}
My problem is that if server does not start to send packets client is behave as the
packet is lost and is starting to send nack to server. (recvfrom is fail when no data is available)I want some advice how can I make difference between those cases , if the server does not start to send the packets and if it sends, but the packet is really lost
You are using UDP. For this protocol its perfectly ok to throw away packets if there is need to do so. So it's not reliable in terms of "what is sent will arrive". What you have to do in your client is to check wether all packets you need arrived, and if not, talk politely to your server to resend those packets you did not receive. To implement this stuff is not that easy,
If you have to use UDP to transfer a largish chunk of data, then design a small application-level protocol that would handle possible packet loss and re-ordering (that's part of what TCP does for you). I would go with something like this:
Datagrams less then MTU (plus IP and UDP headers) in size (say 1024 bytes) to avoid IP fragmentation.
Fixed-length header for each datagram that includes data length and a sequence number, so you can stitch data back together, and detect missed, duplicate, and re-ordered parts.
Acknowledgements from the receiving side of what has been successfully received and put together.
Timeout and retransmission on the sending side when these acks don't come within appropriate time.
you have a loop calling either select() or poll() to determine if data has arrived - if so you then call recvfrom() to read the data.
you can set time out for receive data as follows
ssize_t
recv_timeout(int fd, void *buf, size_t len, int flags)
{
ssize_t ret;
struct timeval tv;
fd_set rset;
// init set
FD_ZERO(&rset);
// add to set
FD_SET(fd, &rset);
// this is set to 60 seconds
tv.tv_sec =
config.idletimeout;
tv.tv_usec = 0;
// NEVER returns before the timeout value.
ret = select(fd, &rset, NULL, NULL, &tv);
if (ret == 0) {
log_message(LOG_INFO,
"Idle Timeout (after select)");
return 0;
} else if (ret < 0) {
log_message(LOG_ERR,
"recv_timeout: select() error \"%s\". Closing connection (fd:%d)",
strerror(errno), fd);
return;
}
ret = recvfrom(fd, buf, len, flags);
return ret;
}
It tells that if there are data ready, Normally, read() should return up to the maximum number of bytes that you've specified, which possibly includes zero bytes (this is actually a valid thing to happen!), but it should never block after previously having reported readiness.
Under Linux, select() may report a socket file descriptor as "ready
for reading", while nevertheless a subsequent read blocks. This could
for example happen when data has arrived but upon examination has
wrong checksum and is discarded. There may be other circumstances in
which a file descriptor is spuriously reported as ready. Thus it may
be safer to use O_NONBLOCK on sockets that should not block.
Look up sliding window protocol here.
The idea is that you divide your payload into packets that fit in a physical udp packet, then number them. You can visualize the buffers as a ring of slots, numbered sequentially in some fashion, e.g. clockwise.
Then you start sending from 12 oclock moving to 1,2,3... In the process, you may (or may not) receive ACK packets from the server that contain the slot number of a packet you sent.
If you receive a ACK, then you can remove that packet from the ring, and place the next unsent packet there which is not already in the ring.
If you receive a NAK for a packet you sent, it means that packet was received by the server with data corruptions, and then you resend it from the ring slot reported in the NAK.
This protocol class allows transmission over channels with data or packet loss (like RS232, UDP, etc). If your underlying data transmission protocol does not provide checksums, then you need to add a checksum for each ring packet you send, so the server can check its integrity, and report back to you.
ACK and NAK packets from the server can also be lost. To handle this, you need to associate a timer with each ring slot, and if you don't receive either a ACK or NAK for a slot when the timer reaches a timeout limit you set, then you retransmit the packet and reset the timer.
Finally, to detect fatal connection loss (i.e. server went down), you can establish a maximum timeout value for all your packets in the ring. To evaluate this, you just count how many consecutive timeouts you have for single slots. If this value exceeds the maximum you have set, then you can consider the connection lost.
Obviously, this protocol class requires dataset assembly on both sides based on packet numbers, since packets may not be sent or received in sequence. The 'ring' helps with this, since packets are removed only after successful transmission, and on the receiving side, only when the previous packet number has already been removed and appended to the growing dataset. However, this is only one strategy, there are others.
Hope this hepls.
As you can probably tell, I'm a bit confused on sliding window with a selective repeat ARQ implementation. If the receiver sends an ACK for a packet, and the ACK gets lost, what does the sender do? Does the sender continue on till the data file with no ACK becomes the bottom of the window and then handle it? Or does the sender wait until the ACK is recieved and then continues?
The server will continue to send data packets until it's window fills up. The receiver will always send a cumulative ACK of the data it has received. This just means that when the receiver sends an ack, it always sends the lowest sequence number it has not received. So if the ack for packet 1 is lost, the server will still send packet 2, the client will ack packet 2 indicating that it is ready to receive packet 3, and the server will update its window with this information upon receiving this ack.
I am programming a gateway which one of the functionality is to destroy connections when enough packets have been exchanged. I would like to know how to properly form RST packets to send to both the client and server to terminate the connection.
To test this, I use ftp connections/sessions. Right now, I am seeing that when I send the RST packets, the client endlessly replies with SYN packets, while the server simply continues the datastream with ACK packets. Note that after I decide to destroy the connection, I block the traffic between both ends.
I am thinking there may be something wrong with the way I handle my SEQ and ACK numbers. I have not been able to find ressources to explain what to do with the SEQ and ACK numbers when sending a RST packet specifically. Right now, I set the SEQ to a new random number (with rand()) and set the ACK to 0 (since I am not using the ACK flag). I invert the source address with destination address and source port with destination port, and have seen that I correctly calculate checksums.
I seems like both the client and server do not accept the termination.
I don't know what 'resources' you are using, but this seems to be completely covered under 'Reset Generation' in section 3.4 of RFC 793. The RST has sequence number zero and the ACK field is set to the incoming ACK field plus the length, etc as described there several times.
When receiving on an ICMP socket, (SOCK_RAW with IPPROTO_ICMP), since
there is no concept of "port" in the ICMP protocol, how can an
application determine that a received packet is not part of some other
TCP/UDP/whatever socket transmission that is also happening at the
same time?
For example, suppose you have an application with 2 threads. Thread 1
sets up a TCP server socket, and continuously receives data from a
connected client. Thread 2 continuously sends echo request packets
(ping) to the same client using an ICMP socket, and then receives echo
replys. What is to prevent Thread 2 from receiving one of the TCP
packets instead?
ICMP is a different protocol from TCP and UDP, as determined by the protocol field in the IP header. When you open a socket with IPPROTO_ICMP, you're telling the socket to transmit and receive only packets with IP headers whose protocol field is set to ICMP.
Similarly, sockets opened with IPPROTO_TCP or IPPROTO_UDP respond only to packets whose IP headers contain a protocol field that is set to TCP or UDP, respectively.
You can check the ICMP header for the type and see if its ICMP Echo Response (Type 0). Also in ICMP, the response will contain the request you had sent in the first place.
Received UDP & TCP packets never passed to raw sockets . IF a process wants to read IP datagram containing UDP or TCP packets the packets must be read at data link layer . check this link
http://aschauf.landshut.org/fh/linux/udp_vs_raw/ch01s03.html
if the packet is not cached at layer 2 then it is processed by kernel .
And if the packet is of icmp protocol and it is of type echo request or timestamp request or address mask request then it is entirely processed by kernel otherwise it will passed to RAW SOCKETS.
Another one all datagrams with a protocol field that the kernel does not understand are passed to raw sockets only basic processing of ip is done on them
At last if datagram arrives in fragments then nothing is passed to raw sockets until all fragments are arived and reassembled .
If you want to learn more, then read this book.