why does TCP over VXLAN in mininet stop sending after switching tunnel? - tcp

topology
This is my experimental setup in Mininet. VM1 and VM2 are separate Virtualbox VM instances running on my computer connected by Bridged adapter, and S1 and S2 are connected with vxlan forwarding.
Then I used D-ITG on H1 and H2 to generate traffic. I send TCP traffic from H1 to H2 and use wireshark to capture. During a 10sec TCP flow, I used a python script that changes the tunnel id of the first rule on S1 from 100 to 200.
If the packet/sec rate and payload size is small enough, the TCP session does not seem to be affected, but when I start sending around 100 packet/sec each with payload of 64 bytes, TCP stop sending after receiving a dup ACK. Here is the wireshark capture:
wireshark1
wireshark2
On the link between H1 and S1 I received ICMP destination unreachable (fragmentation needed).
After the two errors, TCP stopped sending. I understand that the "previous segment not captured" is caused by the fact that when I alter the S1 routing table, there is some down time and packets are dropped by the switch. However, I don't understand why TCP does not initiate retransmission.
This does not happen if I reduce the packet rate or the payload to a smaller amount, or if I use UDP. Is this an issue with the TCP stack, or maybe D-ITG? Or maybe it is an issue with the sequence numbers? Is there a range where if very previous packets are not ACKed, they will not be retransmitted?
This problem has been bothering me for a while, so I hope someone here can maybe provide some clarification. Thanks a lot for reading XD.

I suspected it may be a problem with mininet NICs, so I tried to disable TCP fragmentation offload, and it worked much better. I suppose that the virtual NICs in mininet in a VM could not handle the large amount of traffic generated by D-ITG, so using TCP fragmentation offload can overload? the NIC and cause segmentation errors.
This is just my speculation, but disabling TSO did help my case. Additional input is welcomed!

Related

What are the segments of a packet and parts of a frame in networking?

I have used google for the above questionair but I still couldn't find the answer for the above question. Please help me out on this.
Networking packets are quite a complicated subject, but I will try to explain them to the best of my ability.
Each packet has a source IP and a destination IP, and a body. That’s all it actually needs. Most packets also have a protocol. I don’t know every major protocol, but the basic ones are ICMP, TCP, and UDP(TCP and UDP might be built on ICMP, not sure). Tcp and UDP packets also have a source port and destination port. Using some Linux trickery, you can define your own protocol, but your router probably won’t know what to do with traffic coming in as it isn’t programmed to know whether it should let it in. TCP gives the illusion of a byte stream, but everything is still split into packets. ICMP is just a simple packet, used for pings and similar things. UDP is the most basic of the 3, and is similar to ICMP but with ports, as far as I can tell.
Back to TCP, it splits into multiple packets, because too large of packets are more likely to get lost. TCP also makes sure all packets arrive and in the right order. A stream is nessesary for this, as if you were to try to send your own packet, it wouldn’t have a check for how large, and could get lost very easily if not done right.
A UDP listener simply tells the OS to listen for UDP packets on that port instead of discarding them. When you send a UDP packet, the router remembers the source and destination, and allows the other end to communicate back for a certain length of time.
A TCP listener accepts packets requesting a UDP connection, and sends them to a different port. The router uses a similar strategy to UDP to know if a packet should be let in. Unfortunately, if one side terminates, there is no way for the other side or the router to know. Thus the router will often continue letting in packets to a stream that was closed, which could pose a risk.
This is my understanding, it is very much flawed. Hope I could help nonetheless!

is the UDP or TCP protocol best for sending back un-noticed packets / datagrams

so I'm working on a project where the program can detect when its being scanned for malicious purposes by checking how many ports are being scanned at the same time and scanning them back using the SYN method and I would like to know if the TCP or UDP protocol is better for a so called "counter-scan" to the target without getting noticed I have some ideas like:
I can send them using UDP and the attacker wouldn't notice them .
using the TCP method use the existing 3 way handshake to mask the
SYN packets with his responses
sorry I have no source code since I'm still brain storming
Yes, UDP scan can be done by looking at ICMP (NOT IMCP) port unreachables, but these are often filtered.
I guess UDP would not be less "noticed"--TCP does more harm since it needs state saved (waiting for ACKs).
(nit: please work on your English)

TCP checksum error for fragmented packets

I'm working on a server/client socket application that is using Linux TUN interface.
Server gets packets directly from TUN interface and pass them to clients and clients put received packets directly in the TUN interface.
<Server_TUN---><---Server---><---Clients---><---Client_TUN--->
Sometimes the packets from Server_TUN need to be fragmented in IP layer before transmitting to a client.
So at the server I read a packet from TUN, start fragmenting it in the IP layer and send them via socket to clients.
When the fragmentation logic was implemented, the solution did not work well.
After starting Wireshark on Client_TUN I noticed for all incoming fragmented packets I get TCP Checksum error.
At the given screenshot, frame number 154 is claimed to be reassembled in in 155.
But TCP checksum is claimed to be incorrect!
At server side, I keep tcp data intact and for the given example, while you see the reverse in Wireshark, I've split a packet with 1452 bytes (including IP header) and 30 bytes (Including IP header)
I've also checked the TCP checksum value at the server and its exactly is 0x935e and while I did not think that Checksum offloading matters for incoming packets, I checked offloading at the client and it was off.
$ sudo ethtool -k tun0 | grep ": on"
scatter-gather: on
tx-scatter-gather: on
tx-scatter-gather-fraglist: on
generic-segmentation-offload: on
generic-receive-offload: on
tx-vlan-offload: on
tx-vlan-stag-hw-insert: on
Despite that, because of the solution is not working now, I don't think its caused by offload effect.
Do you have any idea why TCP checksum could be incorrect for fragmented packets?
Hopefully I found the issue. It was my mistake. Some tcp data was missing when I was coping buffers. I was tracing on the indexes and lengths but because of the changes in data, checksum value was calculating differently in the client side.

Packet loss showing at point of entry onto network - what could cause?

A traffic source (server) with a 1gigabit NIC is attached to a 1gigabit port of a Cisco switch.
I mirror this traffic (SPAN) to a separate gigabit port on the same switch and then capture this traffic on a high throughput capture device (riverbed shark).
Wireshark analysis of the capture shows that there is a degree of packet loss - around 0.1% of TCP segments are being lost (based on sequence number analysis).
Given that this is the first point on the network for this traffic, what can cause this loss?
The throughput is not anywhere near 1gigabit, there are no port errors (which might indicate a dodgy patch lead).
In Richard Stevens illustrated TCP book he makes mention of 'local congestion' - where the TCP stack is producing data at a rate faster than the underlying local queues can be emptied.
Could this be what I am seeing?
If so, is there a way to confirm it on an AIX box?
(Stevens example used the Linux 'tc' command for a ppp0 device to demonstrate drops at the lower level)
The lost can be anywhere along the network path.
If there is loss between two hosts, you should be seeing DUP ACKs. You need to see what side is sending the DUP ACKs. This would be the host that isn't receiving all the packets. ( When a packet is not seen, it will send a DUP ACK to ask for the packet again.)
There may be congestion somewhere else along the path. Look for output drops on interfaces. Or CRC erros .

Why do we say the IP protocol in TCP/IP suite is connectionless?

Why is the IP called a connectionless protocol? If so, what is the connection-oriented protocol then?
Thanks.
Update - 1 - 20:21 2010/12/26
I think, to better answer my question, it would be better to explain what "connection" actually means, both physically and logically.
Update - 2 - 9:59 AM 2/1/2013
Based on all the answers below, I come to the feeling that the 'connection' mentioned here should be considered as a set of actions/arrangements/disciplines. Thus it's more an abstract concept rather than a concrete object.
Update - 3 - 11:35 AM 6/18/2015
Here's a more physical explanation:
IP protocol is connectionless in that all packets in IP network are routed independently, they may not necessarily go through the same route, while in a virtual circuit network which is connection oriented, all packets go through the same route. This single route is what 'virtual circuit' means.
With connection, because there's only 1 route, all data packets will arrive in the same order as they are sent out.
Without connection, it is not guaranteed all data packets will arrive
in the same order as they are sent out.
Update - 4 - 9:55 AM 2016/1/20/Wed
One of the characteristics of connection-oriented is that the packet order is preserved. TCP use a sequence number to achieve that but IP has no such facility. Thus TCP is connection-oriented while IP is connection-less.
The basic idea is pretty simple: with IP (on its own -- no TCP, UDP, etc.) you're just sending a packet of data. You simply send some data onto the net with a destination address, but that's it. By itself, IP gives:
no assurance that it'll be delivered
no way to find out if it was
nothing to let the destination know to expect a packet
much of anything else
All it does is specify a minimal packet format so you can get some data from one point to another (e.g., routers know the packet format, so they can look at the destination and send the packet on its next hop).
TCP is connection oriented. Establishing a connection means that at the beginning of a TCP conversation, it does a "three way handshake" so (in particular) the destination knows that a connection with the source has been established. It keeps track of that address internally, so it can/will/does expect more packets from it, and be able to send replies to (for example) acknowledge each packet it receives. The source and destination also cooperate to serial number all the packets for the acknowledgment scheme, so each end knows whether packets it sent were received at the other end. This doesn't involve much physically, but logically it involves allocating some memory on both ends. That includes memory for metadata like the next packet serial number to use, as well as payload data for possible re-transmission until the other side acknowledges receipt of that packet.
TCP/IP means "TCP over IP".
TCP
--
IP
TCP provides the "connection-oriented" logic, ordering and control
IP provides getting packets from A to B however it can: "connectionless"
Notes:
UDP is connection less but at the same level as TCP
Other protocols such as ICMP (used by ping) can run over IP but have nothing to do with TCP
Edit:
"connection-oriented" mean established end to end connection. For example, you pick up the telephone, call someone = you have a connection.
"connection-less" means "send it, see what happens". For example, sending a letter via snail mail.a
So IP gets your packets from A to B, maybe, in any order, not always eventually. TCP sorts them out, acknowledges them, requests a resends and provides the "connection"
Connectionless means that no effort is made to set up a dedicated end-to-end connection, While Connection-Oriented means that when devices communicate, they perform handshaking to set up an end-to-end connection.
IP is an example of the Connectionless protocols , in this kind of protocols you usually send informations in one direction, from source to destination without checking to see if the destination is still there, or if it is prepared to receive the information . Connectionless protocols (Like IP and UDP) are used for example with the Video Conferencing when you don't care if some packets are lost , while you have to use a Connection-Oriented protocol (Like TCP) when you send a File because you want to insure that all the packets are sent successfully (actually we use FTP to transfer Files). Edit :
In telecommunication and computing in
general, a connection is the
successful completion of necessary
arrangements so that two or more
parties (for example, people or
programs) can communicate at a long
distance. In this usage, the term has
a strong physical (hardware)
connotation although logical
(software) elements are usually
involved as well.
The physical connection is layer 1 of
the OSI model, and is the medium
through which the data is transfered.
i.e., cables
The logical connection is layer 3 of
the OSI model, and is the network
portion. Using the Internetwork
Protocol (IP), each host is assigned a
32 bit IP address. e.g. 192.168.1.1
TCP is the connection part of TCP/IP. IP's the addressing.
Or, as an analogy, IP is the address written on the envelope, TCP is the postal system which uses the address as part of the work of getting the envelope from point A to point B.
When two hosts want to communicate using connection oriented protocol, one of them must first initiate a connection and the other must accept it. Logically a connection is made between a port in one host and other port in the other host. Software in one host must perform a connect socket operation, and the other must perform an accept socket operation. Physically the initiator host sends a SYN packet, which contains all four connection identifying numbers (source IP, source port, destination IP, destination port). The other receives it and sends SYN-ACK, the initiator sends an ACK, then the connection are established. After the connection established, then the data could be transferred, in both directions.
In the other hand, connectionless protocol means that we don't need to establish connection to send data. It means the first packet being sent from one host to another could contain data payloads. Of course for upper layer protocols such as UDP, the recipient must be ready first, (e.g.) it must perform a listen udp socket operation.
The connectionless IP became foundation for TCP in the layer above
In TCP, at minimal 2x round trip times are required to send just one packet of data. That is : a->b for SYN, b->a for SYN-ACK, a->b for ACK with DATA, b->a for ACK. For flow rate control, Nagle's algorithm is applied here.
In UDP, only 0.5 round trip times are required : a->b with DATA. But be prepared that some packets could be silently lost and there is no flow control being done. Packets could be sent in the rate that are larger than the capability of the receiving system.
In my knowledge, every layer makes a fool of the one above it. The TCP gets an HTTP message from the Application layer and breaks it into packets. Lets call them data packets. The IP gets these packets one by one from TCP and throws it towards the destination; also, it collects an incoming packet and delivers it to TCP. Now, TCP after sending a packet, waits for an acknowledgement packet from the other side. If it comes, it says the above layer, hey, I have established a connection and now we can communicate! The whole communication process goes on between the TCP layers on both the sides sending and receiving different types of packets with each other (such as data packet, acknowledgement packet, synchronization packet , blah blah packet). It uses other tricks (all packet sending) to ensure the actual data packets to be delivered in ordered as they were broken and assembled. After assembling, it transfers them to the above application layer. That fool thinks that it has got an HTTP message in an established connection but in reality, just packets are being transferred.
I just came across this question today. It was bouncing around in my head all day and didn't make any sense. IP doesn't handle transport. Why would anyone even think of IP as connectionless or connection oriented? It is technically connectionless because it offers no reliability, no guaranteed delivery. But so is my toaster. My toaster offers no guaranteed delivery, so why not call aa toaster connectionless too?
In the end, I found out it's just some stupid title that someone somewhere attached to IP and it stuck, and now everyone calls IP connectionless and has no good reason for it.
Calling IP connectionless implies there is another layer 3 protocol that is connection oriented, but as far as I know, there isn't and it is just plain stupid to specify that IP is connectionless. MAC is connectionless. LLC is connectionless. But that is useless, technically correct info.

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