I'm trying to dissect certain protocol that runs on top of TCP. The messages are sometimes split into different packets - unfortunately I do not have a way of determining the expected length of the message (protocol does not announce this). The only way to determine that the message transmission is complete is to wait for a TCP/IP packet with FIN,ACK flags set.
The issue is that my dissector is not even aware of TCP/IP protocol (nor it should in concern itself with it in reality). Is there any way to reassemble all packets until FIN,ACK is seen?
So the entire TCP connection contains only a single message? You can specify DESEGMENT_UNTIL_FIN in pinfo->desegment_len. See epan/packet_info.h for details.
Related
When will a TCP packet be fragmented at the application layer? When a TCP packet is sent from an application, will the recipient at the application layer ever receive the packet in two or more packets? If so, what conditions cause the packet to be divided. It seems like a packet won't be fragmented until it reaches the Ethernet (at the network layer) limit of 1500 bytes. But, that fragmentation will be transparent to the recipient at the application layer since the network layer will reassemble the fragments before sending the packet up to the next layer, right?
It will be split when it hits a network device with a lower MTU than the packet's size. Most ethernet devices are 1500, but it can often be smaller, like 1492 if that ethernet is going over PPPoE (DSL) because of the extra routing information, even lower if a second layer is added like Windows Internet Connection Sharing. And dialup is normally 576!
In general though you should remember that TCP is not a packet protocol. It uses packets at the lowest level to transmit over IP, but as far as the interface for any TCP stack is concerned, it is a stream protocol and has no requirement to provide you with a 1:1 relationship to the physical packets sent or received (for example most stacks will hold messages until a certain period of time has expired, or there are enough messages to maximize the size of the IP packet for the given MTU)
As an example if you sent two "packets" (call your send function twice), the receiving program might only receive 1 "packet" (the receiving TCP stack might combine them together). If you are implimenting a message type protocol over TCP, you should include a header at the beginning of each message (or some other header/footer mechansim) so that the receiving side can split the TCP stream back into individual messages, either when a message is received in two parts, or when several messages are received as a chunk.
Fragmentation should be transparent to a TCP application. Keep in mind that TCP is a stream protocol: you get a stream of data, not packets! If you are building your application based on the idea of complete data packets then you will have problems unless you add an abstraction layer to assemble whole packets from the stream and then pass the packets up to the application.
The question makes an assumption that is not true -- TCP does not deliver packets to its endpoints, rather, it sends a stream of bytes (octets). If an application writes two strings into TCP, it may be delivered as one string on the other end; likewise, one string may be delivered as two (or more) strings on the other end.
RFC 793, Section 1.5:
"The TCP is able to transfer a
continuous stream of octets in each
direction between its users by
packaging some number of octets into
segments for transmission through the
internet system."
The key words being continuous stream of octets (bytes).
RFC 793, Section 2.8:
"There is no necessary relationship
between push functions and segment
boundaries. The data in any particular
segment may be the result of a single
SEND call, in whole or part, or of
multiple SEND calls."
The entirety of section 2.8 is relevant.
At the application layer there are any number of reasons why the whole 1500 bytes may not show up one read. Various factors in the internal operating system and TCP stack may cause the application to get some bytes in one read call, and some in the next. Yes, the TCP stack has to re-assemble the packet before sending it up, but that doesn't mean your app is going to get it all in one shot (it is LIKELY will get it in one read, but it's not GUARANTEED to get it in one read).
TCP tries to guarantee in-order delivery of bytes, with error checking, automatic re-sends, etc happening behind your back. Think of it as a pipe at the app layer and don't get too bogged down in how the stack actually sends it over the network.
This page is a good source of information about some of the issues that others have brought up, namely the need for data encapsulation on an application protocol by application protocol basis Not quite authoritative in the sense you describe but it has examples and is sourced to some pretty big names in network programming.
If a packet exceeds the maximum MTU of a network device it will be broken up into multiple packets. (Note most equipment is set to 1500 bytes, but this is not a necessity.)
The reconstruction of the packet should be entirely transparent to the applications.
Different network segments can have different MTU values. In that case fragmentation can occur. For more information see TCP Maximum segment size
This (de)fragmentation happens in the TCP layer. In the application layer there are no more packets. TCP presents a contiguous data stream to the application.
A the "application layer" a TCP packet (well, segment really; TCP at its own layer doesn't know from packets) is never fragmented, since it doesn't exist. The application layer is where you see the data as a stream of bytes, delivered reliably and in order.
If you're thinking about it otherwise, you're probably approaching something in the wrong way. However, this is not to say that there might not be a layer above this, say, a sequence of messages delivered over this reliable, in-order bytestream.
Correct - the most informative way to see this is using Wireshark, an invaluable tool. Take the time to figure it out - has saved me several times, and gives a good reality check
If a 3000 byte packet enters an Ethernet network with a default MTU size of 1500 (for ethernet), it will be fragmented into two packets of each 1500 bytes in length. That is the only time I can think of.
Wireshark is your best bet for checking this. I have been using it for a while and am totally impressed
I am writing an application that needs to get access to TCP header fields, for example, a sequence number or a TCP timestamp field.
Is it possible to get sequence numbers (or other header fields) by operating at the socket API without listening on a raw socket? (I want to avoid filtering out all the packets).
I am looking at the TCP_INFO but it has a limited information.
For example, after calling a recvmsg() and getting a data buffer, is it possible to know the sequence number of the segment that delivered the last byte in that received data buffer?
Thanks
You can try to use libpcap to capture packets. This lib allows to specify packet filter using the same syntax as in Wireshark, so you could limit captured packets to one connection only. One downside is that you would have to receive packets in normal way too, what complicated things a bit and is an additional performance overhead.
Update: you can also open raw socket and set Berkeley Packet Filter on it using socket option SO_ATTACH_FILTER. More details are here: https://www.kernel.org/doc/Documentation/networking/filter.txt . However you would have to implement TCP part of IP stack in your code too.
I have a client and a server, where server sends the audio data by RTP packets encapsulated inside UDP. Client receives the packets. As UDP has no flow control, client checks for the sequence number of the packet and rearranges them if they come out of order.
My question here is, I see client never receives packet with some sequence number, as show below in the wireshark -
If this is the case, when i play the audio at client side, it is distorted(obvious). How do i avoid it? What factors effect these? Should i set the socket buffer size to a large value?
Appreciate reply in advance.
EDIT 1: This issue is on QNX platform and not on Linux.
I observed the output of "netstat -p udp" to see if that gives any hint about why packets are getting dropped on QNX and not on Linux.
QNX:
SOCK=/dev/d_usb3/ netstat -p udp
udp:
8673 datagrams received
0 with incomplete header
60 with bad data length field
0 with bad checksum
0 dropped due to no socket
2 broadcast/multicast datagrams dropped due to no socket
0 dropped due to full socket buffers
8611 delivered
8592 PCB hash misses
On Linux I see netstat shows no packet drops with the same server and same audio!
Any leads? Why this might be? Driver issue? Networking stack?
You need to specify how you are handling lost packets in your client.
If you lose packets, that means you have missing data in your audio stream. So your client has to "do something" where it is missing data. Some options are
- play silence (makes a cracking noise due to sharp envelop to 0)
- fade to silence
- estimate waveform by examining adjacent data
- play noise
You play cannot misalign packets or play them with missing packets. For example, suppose you you get packet 1,2,3,4 and 6. You are missing packet 5. You cannot play packet 4 then play packet 6. Something has to happen to fill the space of packet 5.
See this post for more info.
We want to receive packets from udp sockets, the udp packets have variable length and we don't know how long they really are until we receive them (parts of them exactly, length were written in the sixth byte).
We tried the function lrs_set_receive_option with MarkerEnd only to find it has no help on this issue. The reason why we want to receive by packets is that we need to respond some packet by sending back user-defined udp packets.
Is there anybody knows how achieve that?
UPDATE
The LR version seems to be v10 or v11.
We need respond an incoming udp packet by sending back a udp packet immediately.
The udp packet may be like this
| orc code | packet length | Real DATA |
Issue is we can't let loadrunner return data for each packets, sometimes it returns many packets in a buffer, sometimes it waits until timeout though when there has been an incoming packet in the socket buffer. While in the c programming language world, when calling recvfrom(udp socket) we are returned only one udp packet per time (per call) which is want we really want.
If you need raw socket support to intercept at the packet level then you are likely going to have to jump to a DLL virtual user in Visual Studio with the raw socket support.
As to your question on UDP support: Yes, a Winsock user supports both core transport types, UDP and TCP. TCP being the more common variant as connection oriented. However, packet examination is at layer 3 of the OSI model for the carrier protocol IP. The ACK should come before you receive the dataflow for your use in the script. You are looking at assembled data flows in the data.ws when you jump to the TCP and UDP level.
Now, you are likely receiving a warning on receive buffer size mismatch which is taking you down this path with a mismatch to the recording size. There is an easy way to address this. If you take your send buffer and construct it using the lrs_set_send_buffer() function, then anything that returns will be taken as correct, ignoring the previously recorded buffer size and not having to wait for a match or timeout before continuing.
I'm sending many TCP packets each of size 50 bytes over the network. Later I found it out TCP aggregates a few 50 byte packets into a single TCP packet. My question is, is there a way to avoid TCP aggregation in 'C' program?
Packing multiple sent packets into a single TCP packet is handled using an algorithm known as Nagle's algorithm. To disable it, set the TCP_NODELAY option on your socket:
int flag = 1;
setsockopt(fd, IPPROTO_TCP, TCP_NODELAY, &flag, sizeof(flag));
Note that this decreases the efficiency of your network, and should be avoided unless you really do need each packet to be sent immediately.
TCP fundamentally does not provide a packet-based service at the application layer. Applications using TCP that need more than a pure stream must provide their own framing mechanism.
Whilst the TCP_NODELAY option will often cause each application-level write to be carried in a separate frame (unless it's larger than the MTU), there's no guarantee. In particular, if the recieve window has been reduced to zero (common on a connection where you are sending bulk data), then multiple writes can still be coalesced when the receive window opens up again.
Furthermore, even if the sending TCP does not coalesce your writes, there is no way to stop the receiving side from presenting the data from multiple TCP frames in one application-level read from the socket - this will particularly occur when your receiving application is not scheduled for some time, or when packet loss occurs on the connection.
If your application assumes that each application-level write on the sending end will be paired exactly with an application-level read on the receiving end, you are setting yourself up for rare and hard-to-reproduce bugs.
TCP does not, by definition, offer any packetization services. TCP is a byte-stream oriented protocol. The best you can do is to insert your own framing marks into your data. Then if your receiver can at least detect bad frames (aggregated data perhaps) and drop such data, that sort of gets you where you want to be.
The send-as-soon-as-possible approach (ala NODELAY) is not guaranteed to work because your receiver can still aggregate the data. This is more probable if any tcp data gets delayed for whatever reason
One way is to use the TCP_NODELAY flags in the send call. However it depends on the TCP stack if this is supported or not.