ifconfig 1.2.3.4 mtu 1492
This will set MTU to 1492 for incoming, outgoing packets or both? I think it is only for incoming
TLDR: Both. It will only transmit packets with a payload length less than or equal to that size. Similarly, it will only accept packets with a payload length within your MTU. If a device sends a larger packet, it should respond with an ICMP unreachable (oversized) message.
The nitty gritty:
Tuning the MTU for your device is useful because other hops between you and your destination may encapsulate your packet in another form (for example, a VPN or PPPoE.) This layer around your packet results in a bigger packet being sent along the wire. If this new, larger packet exceeds the maximum size of the layer, then the packet will be split into multiple packets (in a perfect world) or will be dropped entirely (in the real world.)
As a practical example, consider having a computer connected over ethernet to an ADSL modem that speaks PPPoE to an ISP. Ethernet allows for a 1500 byte payload, of which 8 bytes will be used by PPPoE. Now we're down to 1492 bytes that can be delivered in a single packet to your ISP. If you were to send a full-size ethernet payload of 1500 bytes, it would get "fragmented" by your router and split into two packets (one with a 1492 byte payload, the other with an 8 byte payload.)
The problem comes when you want to send more data over this connection - lets say you wanted to send 3000 bytes: your computer would split this up based on your MTU - in this case, two packets of 1500 bytes each, and send them to your ADSL modem which would then split them up so that it can fulfill its MTU. Now your 3000 byte data has been fragmented into four packets: two with a payload of 1492 bytes and two with a payload of 8 bytes. This is obviously inefficient, we really only need three packets to send this data. Had your computer been configured with the correct MTU for the network, it would have sent this as three packets in the first place (two 1492 byte packets and one 16 byte packet.)
To avoid this inefficiency, many IP stacks flip a bit in the IP header called "Don't Fragment." In this case, we would have sent our first 1500 byte packet to the ADSL modem and it would have rejected the packet, replying with an Internet Control (ICMP) message informing us that our packet is too large. We then would have retried the transmission with a smaller packet. This is called Path MTU discovery. Similarly, a layer below, at the TCP layer, another factor in avoiding fragmentation is the MSS (Maximum Segment Size) option where both hosts reply with the maximum size packet they can transfer without fragmenting. This is typically computed from the MTU.
The problem here arises when misconfigured firewalls drop all ICMP traffic. When you connect to (say) a web server, you build a TCP session and send that you're willing to accept TCP packets based on your 1500 byte MTU (since you're connected over ethernet to your router.) If the foreign web server wanted to send you a lot of data, they would split this into chunks that (when combined with the TCP and IP headers) came out to 1500 byte payloads and send them to you. Your ISP would receive one of these and then try to wrap it into a PPPoE packet to send to your ADSL modem, but it would be too large to send. So it would reply with an ICMP unreachable, which would (in a perfect world) cause the remote computer to downsize its MSS for the connection and retransmit. If there was a broken firewall in the way, however, this ICMP message would never be reached by the foreign web server and this packet would never make it to you.
Ultimately setting your MTU on your ethernet device is desirable to send the right size frames to your ADSL modem (to avoid it asking you to retransmit with a smaller frame), but it's critical to influence the MSS size you send to remote hosts when building TCP connections.
ifconfig ... mtu <value> sets the MTU for layer2 payloads sent out the interface, and will reject larger layer2 payloads received on this interface. You must ensure your MTU matches on both sides of an ethernet link; you should not have mismatched mtu values anywhere in the same ethernet broadcast domain. Note that the ethernet headers are not included in the MTU you are setting.
Also, ifconfig has not been maintained in linux for ages and is old and deprecated; sadly linux distributions still include it because they're afraid of breaking old scripts. This has the very negative effect of encouraging people to continue using it. You should be using the iproute2 family of commands:
[mpenning#hotcoffee ~]$ sudo ip link set mtu 1492 eth0
[mpenning#hotcoffee ~]$ ip link show eth0
2: eth0: <BROADCAST,MULTICAST,UP,LOWER_UP> mtu 1492 qdisc mq state UP qlen 1000
link/ether 00:1e:c9:cd:46:c8 brd ff:ff:ff:ff:ff:ff
[mpenning#hotcoffee ~]$
Large incoming packets may be dropped based on the interface MTU size.
For example, the default MTU 1500 on
Linux 2.6 CentOS (tested with Ethernet controller: Intel Corporation 80003ES2LAN Gigabit Ethernet Controller (Copper) (rev 01))
drops Jumbo packets >1504. Errors appear in ifconfig and there are rx_long_length_errors indications for this in ethtool -S output.
Increasing MTU indicates Jumbo packets should be supported.
The threshold for when to drop packets based on their size being too large appears to depend on MTU (-4096, -8192, etc.)
Oren
It's the Maximum Transmission Unit, so it definitely sets the outgoing maximum packet size. I'm not sure if will reject incoming packets larger than the MTU.
There is no doubt that MTU configured by ifconfig impacts Tx ip fragmentation, I have no more comments.
But for Rx direction, I find whether the parameter impacts incoming IP packets, it depends. Different manufacturer behaves differently.
I tested all the devices on hand and found 3 cases below.
Test case:
Device0 eth0 (192.168.225.1, mtu 2000)<--ETH cable-->Device1 eth0
(192.168.225.34, mtu MTU_SIZE)
On Device0 ping 192.168.225.34 -s ICMP_SIZE,
Checking how MTU_SIZE impacts Rx of Device1.
case 1:
Device1 = Linux 4.4.0 with Intel I218-LM:
When MTU_SIZE=1500, ping succeeds at ICMP_SIZE=1476, fails at ICMP_SIZE=1477 and above. It seems that there is a PRACTICAL MTU=1504 (20B(IP header)+8B(ICMP header)+1476B(ICMP data)).
When MTU_SIZE=1490, ping succeeds at ICMP_SIZE=1476, fails at ICMP_SIZE=1477 and above, behave the same as MTU_SIZE=1500.
When MTU_SIZE=1501, ping succeeds at ICMP_SIZE=1476, 1478, 1600, 1900. It seems that jumbo frame is switched on once MTU_SIZE is set >1500 and there is no 1504 restriction any more.
case 2:
Device1 = Linux 3.18.31 with Qualcomm Atheros AR8151 v2.0 Gigabit Ethernet:
When MTU_SIZE=1500, ping succeeds at ICMP_SIZE=1476, fails at ICMP_SIZE=1477 and above.
When MTU_SIZE=1490, ping succeeds at ICMP_SIZE=1466, fails at ICMP_SIZE=1467 and above.
When MTU_SIZE=1501, ping succeeds at ICMP_SIZE=1477, fails at ICMP_SIZE=1478 and above.
When MTU_SIZE=500, ping succeeds at ICMP_SIZE=476, fails at ICMP_SIZE=477 and above.
When MTU_SIZE=1900, ping succeeds at ICMP_SIZE=1876, fails at ICMP_SIZE=1877 and above.
This case behaves exactly as Edward Thomson said, except that in my test the PRACTICAL MTU=MTU_SIZE+4.
case 3:
Device1 = Linux 4.4.50 with Raspberry Pi 2 Module B ETH:
When MTU_SIZE=1500, ping succeeds at ICMP_SIZE=1472, fails at ICMP_SIZE=1473 and above. So there is a PRACTICAL MTU=1500 (20B(IP header)+8B(ICMP header)+1472B(ICMP data)) working there.
When MTU_SIZE=1490, behave the same as MTU_SIZE=1500.
When MTU_SIZE=1501, behave the same as MTU_SIZE=1500.
When MTU_SIZE=2000, behave the same as MTU_SIZE=1500.
When MTU_SIZE=500, behave the same as MTU_SIZE=1500.
This case behaves exactly as Ron Maupin said in Why MTU configuration doesn't take effect on receiving direction?.
To sum it all, in real world, after you set ifconfig mtu,
sometimes the Rx IP packts get dropped when exceed 1504 , no matter what MTU value you set (except that the jumbo frame is enabled).
sometimes the Rx IP packts get dropped when exceed the MTU+4 you set on receiving device.
sometimes the Rx IP packts get dropped when exceed 1500, no matter what MTU value you set.
... ...
Related
I'm trying to pentest some IPSEC implementation for a uni project, and following this guide I'm stuck at:
Step 1 (common): Forging an ICMP PTB packet from the untrusted network The attacker first has to forge an appropriate ICMP PTB packet (a single packet is sufficient). This is done by eavesdropping a valid packet from the IPsec tunnel on the untrusted network. Then the attacker forges an ICMP PTB packet, specifying a very small MTU value equal or smaller than 576 with IPv4 (resp. 1280 with IPv6). The attacker can use 0 for instance. This packet spoofs the IP address of a router of the untrusted network (in case the source IP address is checked), and in order to bypass the IPsec protection mechanism against blind attacks, it includes as a payload a part of the outer IP packet that has just been eavesdropped. This is the only packet an attacker needs to send. None of the following steps involve the attacker.
I know what MTU is, but what does the bold statement mean?
How do I set the MTU size of a packet with scapy?
It means that I have to set the size of a IP packet less than 576 bytes?
It's already set to 140 B,at least it shows this with len command.
There's something that I didn't get right, maybe I have to set the fragmentation?
I know nothing about the subject, but some quick searching seems to indicate that it's referring to an IPv6 ICMP packet with a type of 2 ("packet too big").
Then from some poking around scapy, this appears to be how you'd create one:
from scapy.layers.inet6 import ICMPv6PacketTooBig
icmp_ptb = ICMPv6PacketTooBig(mtu=0)
Of course though, you'll need to do some testing to verify this.
I'm doing some UDP bandwidth tests using iperf (https://iperf.fr/) over IPv6. I have very bad results when using a Linux UDP client with the following command line:
iperf -u -V -c fe80::5910:d4ff:fe31:5b10%usb0 -b 100m
Investigating the issue with Wireshark I have seen there is some fragmentation while the client is sending data. To be more precise, I see UDP client outgoing packets with size 1510 bytes and 92 bytes, alternating.
For example, the UDP packets that I see have the following pattern (in size): 1510, 92, 1510, 92, 1510, 92,...,1510, 92,...
Reading iperf2 documentation I read the following for option (-l) :
The length of buffers to read or write. iPerf works by writing an array of len bytes a number of times. Default is 8 KB for TCP, 1470 bytes for UDP. Note for UDP, this is the datagram size and needs to be lowered when using IPv6 addressing to 1450 or less to avoid fragmentation. See also the -n and -t options.
I have tried to do the same bandwidth test by replacing the Linux iperf UDP client command line with the following:
iperf -u -V -c fe80::5910:d4ff:fe31:5b10%usb0 -b 100m -l1450
and I see good results. Looking at the Wireshark capture I see no fragmentation anymore.
Doing the same test over IPv4 I don't need to change the default UDP datagram size (I don't need to use '-l' option) to get good results.
So my conclusion is that fragmentation (over IPv6) is responsible for poor bandwidth performances.
Anyway, I'm wondering what really happens when setting UDP datagram size to 1450 over IPv6. Why do I have fragmentation over IPv6 and not over IPv4 with default value for UDP datagram size? Moreover, why do I have no fragmentation when reducing the UDP datagram size to 1450?
Thank you.
The base IPv4 header is 20 bytes, the base IPv6 header is 40 bytes and the UDP header is 8 bytes.
With IPv4 the total packet size is 1470+8+20=1498, which is less than the default ethernet MTU of 1500.
With IPv6 the total packet size is 1470+8+40=1518, which is more than 1500 and has to be fragmented.
Now let's look into your observations. You see packets of size 1510 and 92. Those include the ethernet header, which is 14 bytes. Your IPv6 packets are therefore 1496 and 78 bytes. The contents of the big packet are: IPv6 header (40 bytes), a fragmentation header (8), the UDP header (8) and 1440 bytes of data. The smaller packet contains the IPv6 header (40), a fragmentation header (8) and the remaining 30 bytes of data.
The most common MTU for Ethernet is 1500, not including ethernet frame headers. This means you can send 1500 bytes in one packet over the wire, including IP headers. IPv6 headers are larger than IPv4 headers for several reasons, with the most important being that IPv6 addresses are larger than IPv4. So when you run with the default value over IPv6 your packet size goes over the MTU size and the packet needs to be split into two; a procedure known as fragmentation.
I'm exploring/benchmarking various IPC mechanisms for low latency communication between two processes in the same system. I'm using RHEL 6 system for benchmarking.
I'm currently looking into socket based communication through loopback. Since it is the loopback device, the packets do not even hit the NIC. Instead, the loopback linux driver loopbacks the packets to the destination.
However looking into the results of netstat -i, I see an MTU defined for the loopback. What's the role of this and the potential impact on bandwidth?
Name Mtu Network Address Ipkts Ierrs Opkts Oerrs Coll
lo0 16384 localhost ::1 1738 - 1738 - -
loopback isn't a physical interface, but still the tcp/ip stack runs a lot of operations on it.
To increase performances on local transfers, kernel developers bumped its mtu from 16 Kb to 64 Kb.
See this commit in the Linux kernel and its rationale:
loopback current mtu of 16436 bytes allows no more than 3 MSS TCP
segments per frame, or 48 Kbytes. Changing mtu to 64K allows TCP
stack to build large frames and significantly reduces stack overhead.
Performance boost on bulk TCP transferts can be up to 30 %, partly because we now have one ACK message for two 64KB segments, and a lower
probability of hitting /proc/sys/net/ipv4/tcp_reordering default limit.
--- a/drivers/net/loopback.c
+++ b/drivers/net/loopback.c
static void loopback_setup(struct net_device *dev)
{
- dev->mtu = (16 * 1024) + 20 + 20 + 12;
+ dev->mtu = 64 * 1024;
I have C application which transmits UDP stream. It works well in most of servers, but its crazy on few servers.
I have 100 Mbps network connection say eth1 on server. Using this network I usually transmit (TX) around 10-30 Mbps UDP streams, and this network connection will have around 100-300 Kbps RX to server. I have other network connection say eth0 in server from which C application receives UDP streams and forwards to 100 Mbps network connection, eth1.
My application uses blocking sendto() function to transmit UDP packets in eth1. Packets are of variable length, from 17 bytes to maximum 1333 bytes. But most of time, more than 1000 bytes.
The problem is: sometime sendto function blocks on eth1 for huge time around 1 second. This happens once in every 30 seconds to 3 minutes. When sendto blocks, I will have lot of UDP packets buffered in UDP receive buffer from eth0 by kernel, from where C application receive packets. Once sendto returns from long blocking call on eth1, C application will have lot of buffered packets to transmit from eth0. And then C application transmits all these buffered packets with next sendto calls. This will create spike in rate at other endpoint which receives UDP stream from eth1. This will create Z like rate graph at other endpoint. So this Z like spike in rate is my problem.
I have tried to increase wmem_default from around 131 KB to 5 MB in kernel setting to overcome spike. And setting this resolves my issue of spike. Now I don't get Z like spike in rate at other endpoint, but I got new issue. The new issue is: I get lot of packet losses in place of spike. I think it may be due to send buffer of eth1 accumulating lot of packets to send while sending current packet from eth1 takes lot of time (this is why may be sendto blocking long). And at next instant when NIC sends all accumulated packets from send buffer in short time, this may be causing network congestion and I may be getting lot of packet losses instead of spike.
So, this is second problem. But I think root cause is: why sometime NIC pauses for long time while sending traffic, once in every 30 seconds to 3 minutes?
May be I need to look in TX ring buffer of driver of eth1? When socket send buffer gets full due to NIC not transmitting all in time (due to random long TX pauses), then next call to sendto blocks for room in socket send buffer, does that also blocks for room in driver TX ring buffer?
Please dont tell me that UDP is unreliable and we can't control packet losses. I know that its unreliable and UDP packets can be lost. But I am sure still we can do something to minimize packet losses.
EDIT
I have tried to increase wmem_default from around 131 KB to 5 MB in kernel setting to overcome spike. And also I have removed blocking sendto call. Now I use like: sendto(sockfd, buf, len, MSG_DONTWAIT ,dest_addr, addrlen); with large send buffer using wmem_default. Also I am not getting any EAGAIN or EWOULDBLOCK errors on sendto due to large send buffer, but still packets loosing in place of spike.
EDIT
As non-blocking sendto call with huge wmem_default, and as NO any EAGAIN or EWOULDBLOCK errors from sendto, spikes have been removed because no much packets accumulating in receive buffer of eth0. I think its possible solution from application side. But main problem is why NIC slows every few moments? What can be possible reasons? While it resumes from long TX pause, and may be it will have lot of packets accumulated in send buffer, which will be sent as burst next moment and congesting network so lot of packet losses.
More update
I use same this C application to transmit locally in machine (127.0.0.1), and I never get any spikes or packet losses problems locally.
The problem is: sometime sendto function blocks on eth1 for huge time around 1 second.
Blocking sendto may block, surprisingly.
The problem is: sometime sendto function blocks on eth1 for huge time around 1 second.
It could be that IP stack is performing path MTU discovery:
While MTU discovery is in progress, initial packets from datagram sockets may be dropped. Applications using UDP should be aware of this and not take it into account for their packet retransmit strategy.
I have tried to increase wmem_default from around 131 KB to 5 MB in kernel setting to overcome spike.
Be careful with increasing buffer sizes. After a certain limit increasing buffer sizes only increases the amount of queuing and hence delay, leading to the infamous bufferbloat.
You may also play around with NIC Queuing Disciplines, they are responsible for dropping outgoing packets.
I'm trying to understand some behavior I'm seeing in the context of sending UDP packets.
I have two little Java programs: one that transmits UDP packets, and the other that receives them. I'm running them locally on my network between two computers that are connected via a single switch.
The MTU setting (reported by /sbin/ifconfig) is 1500 on both network adapters.
If I send packets with a size < 1500, I receive them. Expected.
If I send packets with 1500 < size < 24258 I receive them. Expected. I have confirmed via wireshark that the IP layer is fragmenting them.
If I send packets with size > 24258, they are lost. Not Expected. When I run wireshark on the receiving side, I don't see any of these packets.
I was able to see similar behavior with ping -s.
ping -s 24258 hostA works but
ping -s 24259 hostA fails.
Does anyone understand what may be happening, or have ideas of what I should be looking for?
Both computers are running CentOS 5 64-bit. I'm using a 1.6 JDK, but I don't really think it's a programming problem, it's a networking or maybe OS problem.
Implementations of the IP protocol are not required to be capable of handling arbitrarily large packets. In theory, the maximum possible IP packet size is 65,535 octets, but the standard only requires that implementations support at least 576 octets.
It would appear that your host's implementation supports a maximum size much greater than 576, but still significantly smaller than the maximum theoretical size of 65,535. (I don't think the switch should be a problem, because it shouldn't need to do any defragmentation -- it's not even operating at the IP layer).
The IP standard further recommends that hosts not send packets larger than 576 bytes, unless they are certain that the receiving host can handle the larger packet size. You should maybe consider whether or not it would be better for your program to send a smaller packet size. 24,529 seems awfully large to me. I think there may be a possibility that a lot of hosts won't handle packets that large.
Note that these packet size limits are entirely separate from MTU (the maximum frame size supported by the data link layer protocol).
I found the following which may be of interest:
Determine the maximum size of a UDP datagram packet on Linux
Set the DF bit in the IP header and send continually larger packets to determine at what point a packet is fragmented as per Path MTU Discovery. Packet fragmentation should then result in a ICMP type 3 packet with code 4 indicating that the packet was too large to be sent without being fragmented.
Dan's answer is useful but note that after headers you're really limited to 65507 bytes.