I'm implementing TCP in Objective C and C.
When I send a Syn Packet to a server I do not get an answer.
A pcap file of the packet can be found here: Tcp-Syn.pcap
Is the packet malformed or am I missing some convention which leads to my packet being dropped?
Open your file in wireshark.
Go to Edit->Preferences->Protocols->TCP , enable "Validate the TCP checksum if possible"
You will find that the TCP checksum you've generated is wrong.
The MAC addresses in the ethernet headers are all 0 as well, which looks odd - where's this packet going to ?
Related
I'm working on a Win32 application and have noticed I am getting checksum issues in Wireshark (checking is not enabled by default). The IP header checksum is being set to 0 (this is not a massive issue but it would be preferable to sort it) and the UDP checksum is being set to the value that is expected for the IP header checksum.
I am not doing anything to set the IP header checksum since it is out of scope for my project, but I have tried the steps here (also for the UDP checksum) to disable checksum offloading but that has not made a difference. No idea how to get it to set a value for the IP header checksum.
Regarding the UDP checksum, something I think I can affect, I have been experimenting with setsockopt, mainly setsockopt(UdpSocket, SOL_SOCKET, UDP_CHECKSUM_COVERAGE, (const char*)&t, sizeof(t)), where t is a bool/DWORD set to true, and the setsockopts line is after socket() and bind(), before sendTo(). I have also tried changing SOL_SOCKET to IPPROTO_UDP. I know the socket is set up correctly since it has all been working fine before looking at the checksums. Does anyone know any other things I can try? I feel like I am out of options.
Thanks for any help you can give!
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.
I am sorry if the question is too naive, but I am confused. I want to send IPv6 jumbograms (to be able to multicast packets of size > 64 KB). I have been able to multicast normal IPv6 UDP packets successfully.
For sending jumbograms, from RFC 2675, I get that I have to make the following changes :
set payload length to 0
set next header to hop-by-hop
But, I don't get how to implement these in c socket programming (which function calls to make etc.). Do I have to create a custom header or are there functions like sendto available to send jumbograms?
You could use raw sockets if you are making your own headers. For more information, type man -s7 raw or look here. Note you will effectively need to implement your own IP stack that way.
However, my understanding is that linux itself supports IPv6 jumbograms natively so you don't need to bother. Try ifconfig lo mtu 100000 and do some tests over the loopback device to check.
I suspect the issue might be that your network adaptor and everything on the path (end to end) needs to support the jumbograms too.
When receiving a UDP packet, where I do not know neither content nor structure, how can I find out what the content is? Is that possible somehow? Wireshark tells me the data is:
12:03:01:00:00:00:00:00:1f:44:fe:a2:07:e2:4c:28:00:15:e1:e0:00:80:12:61
The question is just about the data itself, not header of lower layers.
Thanks in advance for your help.
Wireshark shows UDP header. There is 2 port numbers. Usually another port number is reserved for the used protocol (not always).
You may look from the port reservation table which is the used protocol.
And if you are lucky, you find the protocol specification from Web and you can open the content of the packet.
I'm programming an offline packets decoding program in C under Windows 7 x86.
I wonder how it is possible to know packet protocol, either if it is UDP or TCP?
You can know by checking the IP packet header, there is a Protocol field in the packet header that is used to indicate the type of the packet according to its value :
1 is ICMP
6 is TCP
17 is UDP
and so on. More information on this is available on Wikipedia
Edit: Here's the list of all the possible values for that field.
P.S:
I'm assuming IPv4 here, I don't know if things are the same with IPv6
The protocol is available in the IP header. Read more here