On a Linux router I wrote a C-program which uses pcap to get the IP header, and length of the packet. In that way I am able to gather statistics and measure bandwidth based on IP. Pretty neat. :-)
Now the traffic and number of users has grown, and the old program starts to struggle. That is, the router struggles to cope with the massive amount of packets. It's over 50000 packets per second all in all in "prime time".
The program itself is pretty optimized. I don't want to show off, but I believe it's as good as it can get. It reads the IP header, and the packet length. It then converts the IP to a index (just a simple subtract), and the length of the packet is stored (accumulated) in an array. Every now and then (actually a SIGALRM) it stores the array in a MySQL database.
My question is: Is there any other way to tap into an ethernet device to get the bit-stream "cheaper" than pcap?
I can of course modify the ethernet driver to include single IP statistics gathering, but that seems a little overkill.
Basically my program is a 'tcpdump' on a busy eth0 and that will eventually kill my router.
Have you considered PF_RING? It's still the pcap-like world, but on steroids - thanks to the zero-copy mechanism:
As you see, there is a kernel module that provides low-level packet copying into the PF_RING buffer, and there is the userland part that allows to access this buffer.
Who needs PF_RING?
Basically everyone who has to handle many packets per second. The term ‘many’ changes according to the hardware you use for traffic analysis. It can range from 80k pkt/sec on a 1,2GHz ARM to 14M pkt/sec and above on a low-end 2,5GHz Xeon. PF_RING not only enables you to capture packets faster, it also captures packets more efficiently preserving CPU cycles....
I highly recommend you to use PF_RING ZC. It could be found under /userland/examples_zc. it is part of pf_ring.
you can handle and capture tens of Gbps traffics in line rate by pf_ring zc.
Related
I have a simple application which receives packets of fixed ethertype via raw socket (the transport is ethernet), and sends two duplicates over another interface (via raw socket):
recvfrom() //blocking
//make duplicate
//add tail
sendto(packet1);
sendto(packet2);
I want two increase throughput. I need at least 4000 frames/second, can't change packet size. How can I achieve these? The system is embedded (AM335x SoC), kernel is 4.14.40... How can I encrease the performance?
A few considerations:
I know you said you can't change "packet size", but using buffered writes and infrequent flushes might really help performance
You could enable jumbo frames
You could disable Nagle
Usually people don't want to change their packet sizes, because they expect a 1-to-1 correspondence between their send()'s and recv()'s. This is not a good thing, because TCP specifically does not ensure that your send()'s and recv()'s will have a 1-1 correspondence. They usually will be 1-1, but they are not guaranteed to do so. Transmitting data with Nagle enabled or over many router hops or without Path MTU Discovery enabled makes the 1-1 relationship less likely.
So if you use buffering, and frame your data somehow (EG 1: terminate messages with a nul byte, if your data cannot otherwise have a nul or EG 2: transfer lengths as network shorts or something, so you know how much to read), you'll likely be killing two birds with one stone - that is, you'll get better speed and better reliability.
I am developing a program that sniffs network packets using a raw socket (AF_PACKET, SOCK_RAW) and processes them in some way.
I am not sure whether my program runs fast enough and succeeds to capture all packets on the socket. I am worried that the recieve buffer for this socket occainally gets full (due to traffic bursts) and some packets are dropped.
How do I know if packets were dropped due to lack of space in the
socket's receive buffer?
I have tried running ss -f link -nlp.
This outputs the number of bytes that are currently stored in the revice buffer for that socket, but I can not tell if any packets were dropped.
I am using Ubuntu 14.04.2 LTS (GNU/Linux 3.13.0-52-generic x86_64).
Thanks.
I was having a similar problem as you. I knew that tcpdump was able to to generate statistics about packet drops, so I tried to figure out how it did that. By looking at the code of tcpdump, I noticed that it is not generating those statistic by itself, but that it is using the libpcap library to get those statistics. The libpcap is on the other hand getting those statistics by accessing the if_packet.h header and calling the PACKET_STATISTICS socket option (at least I think so, but I'm no C expert).
Therefore, I saw only two solutions to the problem:
I had to interact somehow with the linux header files from my Pyhton script to get the packet statistics, which seemed a bit complicated.
Use the Python version of libpcap which is pypcap to get those information.
Since I had no clue how to do the first thing, I implemented the second option. Here is an example how to get packet statistics using pypcap and how to get the packet data using dpkg:
import pcap
import dpkt
import socket
pc=pcap.pcap(name="eth0", timeout_ms=10000, immediate=True)
def packet_handler(ts,pkt):
#printing packet statistic (packets received, packets dropped, packets dropped by interface
print pc.stats()
#example packet parsing using dpkt
eth=dpkt.ethernet.Ethernet(pkt)
if eth.type != dpkt.ethernet.ETH_TYPE_IP:
return
ip =eth.data
layer4=ip.data
ipsrc=socket.inet_ntoa(ip.src)
ipdst=socket.inet_ntoa(ip.dst)
pc.loop(0,packet_handler)
tpacket_stats structure is defined in linux/packet.h header file
Create variable using the tpacket_stats structre and pass it to getSockOpt with PACKET_STATISTICS SOL_SOCKET options will give packets received and dropped count.
-- some times drop can be due to buffer size
-- so if you want to decrease the drop count check increasing the buffersize using setsockopt function
First off, switch your operating system.
You need a reliable, network oriented operating system. Not some pink fluffy "ease of use" with "security" functionality enabled. NetBSD or Gentoo/ArchLinux (the bare installations, not the GUI kitted ones).
Start a simultaneous tcpdump on a network tap and capture the traffic you're supposed to receive along side of your program and compare the results.
There's no efficient way to check if you've received all the packets you intended to on the receiving end since the packets might be dropped on a lower level than you anticipate.
Also this is a question for Unix # StackOverflow, there's no programming here what I can see, at least there's no code.
The only certain way to verify packet drops is to have a much more beefy sender (perhaps a farm of machines that send packets) to a single client, record every packet sent to your reciever. Have the statistical data analyzed and compared against your senders and see how much you dropped.
The cheaper way is to buy a network tap or even more ad-hoc enable port mirroring in your switch if possible. This enables you to dump as much traffic as possible into a second machine.
This will give you a more accurate result because your application machine will be busy as it is taking care of incoming traffic and processing it.
Further more, this is why network taps are effective because they split the communication up into two channels, the receiving and sending directions of your traffic if you will. This enables you to capture traffic on two separate machines (also using tcpdump, but instead of a mirrored port, you get a more accurate traffic mirroring).
So either use port mirroring
Or you buy one of these:
I want to scan the WiFi on b/g interface, and I want to sniff packets on each channel, by spending 100 ms on each channel. One of the biggest requirements I have is not to store the packets I get (because of less disk space), my application will parse the packets, retrieve Tx MAC and RSSI, and would construct the list (MAC, Avg RSSI, #Records) at the end of every minute, and then clear this list and start over again.
I've figured out two ways to do channel hop on linux:
Option 1: Use wi_set_channel(struct wif *, channel number) system call in C, and write the code in C to sniff all the packets
Option 2: Use linux command iw dev wlan0 set channel 4, and use any language like python+scapy OR C to sniff the packets
I'd like to know which is more efficient of the two, if at all, so that the delay/wait for WiFi interface to switch to a different channel is minimal. I suspect that this delay would mean loss of packet while the switch to a different channel happens, is that the case?
I would also like to know some of the other ways to solve this problem in linux.
Answer to your first question us straight forward, use Option1 and have two threads doing the work - one thread populating an in-memory circular buffer with packets collected from channels and second thread processing them in sequence. You can determine best packet discarding algo depending on the measured performance of processing thread and other factors if any.
As for the second question, I would go with the above for being in complete control on exactly how you can tune the algorithm rather than depending on canned processing tools.
When capturing network traffic for debugging, there seem to be two common approaches:
Use a raw socket.
Use libpcap.
Performance-wise, is there much difference between these two approaches? libpcap seems a nice compatible way to listen to a real network connection or to replay some canned data, but does that feature set come with a performance hit?
The answer is intended to explain more about the libpcap.
libpcap uses the PF_PACKET to capture packets on an interface. Refer to the following link.
https://www.kernel.org/doc/Documentation/networking/packet_mmap.txt
From the above link
In Linux 2.4/2.6/3.x if PACKET_MMAP is not enabled, the capture process is very
inefficient. It uses very limited buffers and requires one system call to
capture each packet, it requires two if you want to get packet's timestamp
(like libpcap always does).
In the other hand PACKET_MMAP is very efficient. PACKET_MMAP provides a size
configurable circular buffer mapped in user space that can be used to either
send or receive packets. This way reading packets just needs to wait for them,
most of the time there is no need to issue a single system call. Concerning
transmission, multiple packets can be sent through one system call to get the
highest bandwidth. By using a shared buffer between the kernel and the user
also has the benefit of minimizing packet copies.
performance improvement may vary depending on PF_PACKET implementation is used.
From https://www.kernel.org/doc/Documentation/networking/packet_mmap.txt -
It is said that TPACKET_V3 brings the following benefits:
*) ~15 - 20% reduction in CPU-usage
*) ~20% increase in packet capture rate
The downside of using libpcap -
If an application needs to hold the packet then it may need to make
a copy of the incoming packet.
Refer to manpage of pcap_next_ex.
pcap_next_ex() reads the next packet and returns a success/failure indication. If the packet was read without problems, the pointer
pointed to by the pkt_header argument is set to point to the
pcap_pkthdr struct for the packet, and the pointer pointed to by the
pkt_data argument is set to point to the data in the packet. The
struct pcap_pkthdr and the packet data are not to be freed by the
caller, and are not guaranteed to be valid after the next call to
pcap_next_ex(), pcap_next(), pcap_loop(), or pcap_dispatch(); if the
code needs them to remain valid, it must make a copy of them.
Performance penalty if application only interested in incoming
packets.
PF_PACKET works as taps in the kernel i.e. all the incoming and outgoing packets are delivered to PF_SOCKET. Which results in an expensive call to packet_rcv for all the outgoing packets. Since libpcap uses the PF_PACKET, so libpcap can capture all the incoming as well outgoing packets.
if application is only interested in incoming packets then outgoing packets can be discarded by setting pcap_setdirection on the libpcap handle. libpcap internally discards the outgoing packets by checking the flags on the packet metadata.
So in essence, outgoing packets are still seen by the libpcap but only to be discarded later. This is performance penalty for the application which is interested in incoming packets only.
Raw packet works on IP level (OSI layer 3), pcap on data link layer (OSI layer 2). So its less a performance issue and more a question of what you want to capture. If performance is your main issue search for PF_RING etc, that's what current IDS use for capturing.
Edit: raw packets can be either IP level (AF_INET) or data link layer (AF_PACKET), pcap might actually use raw sockets, see Does libpcap use raw sockets underneath them?
I need to gather some network statistic to test my server application. I've tried many linux tools, but nothing I've found suits my needs.
Basically I want to gather some UDP statistics (bytes/time_interval, packets/time_interval, packets_loss), but regarding only two particular hosts - for example I want to get UDP statistic from traffic going from IP_A:PORT_A to IP_B:PORT_B.
Tools like tcpdump/wireshark can easily dump such traffic but I have problems with getting statistics like temporary speed (too see throughput peeks), and linux system statistics gives me number for all traffic.
It would be better to get text output so it will be possible to parse it.
Anyone has any idea how can I achieve it?
Thanks in advance
Harnen
Here's a tutorial for the libpcap library:
http://www.systhread.net/texts/200805lpcap1.php
To determine packets lost, your program will probably want to work on a pair of logs, and make sure UDP messages on the source are found on the destination. A good method for doing this is to maintain a window of packets equal to the amount of time your timeout is set to, load all the packets into the window, sort them, then search for all the packets in the desired time frame, marking them as found as you go. Once you've exhausted a minute, remove half of that minute from the buffer, and load the next thirty seconds and re-sort.
If you have lots (millions? probably should profile it) of packets, it may be faster to use what's called a Counting Bloom Filter, so you can determine if your packet is "probably" in there very quickly.
If you weren't looking for programming advice, take your question to serverfault.