In networking, what is the relationship between the number of bits and a packet? In other words, how many bits are there in a packet? Is this number constant? If not, what does it depend on? (protocols, physical constraints...)
Thanx.
Each level of the TCP layer has different terminology for the "packet":
TCP - Segment
IP - Datagram
Data Link layer - Frame
Physical - bits or more generic a packet
I am assuming you are asking about what eventually goes into the network.
So, the size of a frame is influenced by the Maximum Transmission Unit (MTU), ie the max no of bits/packet your channel supports (channel here means multiple links/hops, MTU of the entire path is equal to the minimum of the MTUs of each hop in the path).
Since, it varies, so your packet/bit can't be constant. There are protocols employed that determine the path MTU called Path MTU Discovery using ICMP message.
Read more:
RFC 1191 (details how MTU for the route is determined): https://www.rfc-editor.org/rfc/rfc1191
MTU on wikipedia: http://en.wikipedia.org/wiki/Maximum_transmission_unit
Related
We all know that TCP is a streaming protocol.
Unlike UDP, which guarantees that the whole client message will be delivered to server as a single datagram (we're talking about transport layer, so avoid MTU at the moment), TCP can fragment one "message" (data that was passed to the send function) into several smaller packets so we have to use our own delimiters as a messages' borders.
The question is -- how TCP chooses which data should be fragmented and by which size? Is there any common / standard methods that it uses?
TCP uses a parameter called Maximum Segment Size:
The maximum segment size (MSS) is a parameter of the options field of the TCP header that specifies the largest amount of data, specified in bytes, that a computer or communications device can receive in a single TCP segment. It does not count the TCP header or the IP header. The IP datagram containing a TCP segment may be self-contained within a single packet, or it may be reconstructed from several fragmented pieces; either way, the MSS limit applies to the total amount of data contained in the final, reconstructed TCP segment.
The default TCP Maximum Segment Size is 536. Where a host wishes to set the maximum segment size to a value other than the default, the maximum segment size is specified as a TCP option, initially in the TCP SYN packet during the TCP handshake. The value cannot be changed after the connection is established.
If path MTU discovery is enabled, MSS is set to that minus TCP headers size.
On Linux TCP_MAXSEG socket option controls the parameter:
if this option is set before connection establishment, it also changes the MSS value announced to the other end in the initial packet. Values greater than the (eventual) interface MTU have no effect. TCP will also impose its minimum and maximum bounds over the value provided.
As we all perfectly know, UDP does not support retransmission along with some other things.
We also aware of such thing like MTU that works basically in the following way -- when one of the network devices on the path between source and destination points does not support packet of some size, it just drops it.
In case of TCP, it's not a problem -- it already knows MSS after handshake that is always less than MTU (am I right?), so there's no possibility to send a packet with the size greater than MTU.
However, I wonder how does it work in case of UDP? As I already said, there's no retransmission in this protocol and there's no such thing like MSS. So what happens when the packet is dropped due to exceeding MTU?
Or it just works because of the MTU nature (it actually belongs to the IP layer, not the transport layer protocols like UDP or TCP)? So the IP layer reconstruct the dropped packet in smaller units and send it again?
First of all, you must distinguish between the local MTU, which is just the MTU of the local link, and the path MTU (PMTU), which is the smallest MTU of the local link. Consider the following topology:
1500 1480 1500
A -------- B -------- C -------- D
then A's local MTU is 1500, but the PMTU is just 1480.
When router B receives a packet of size 1500 which it needs to forward, and the DF bit is set, it sends an ICMP packet back to the sender with the next hop's MTU, 1480 in this case. The sender can then reduce the packet size.
In TCP, this is done transparently by the network stack. In UDP, the application needs to deal with it. There are three ways to do that:
always send packets that are small enough; 1024 is always safe over IPv6, and 512 is usually (but not always) safe over IPv4;
use a connected UDP socket, and react to an EMSGSIZE error by reducing the packet size; or
use any kind of UDP socket, request the PMTU ancillary data, and use the data provided.
Technique (3) is the most efficient. For IPv6, it is described in Section 11.3 of RFC 3542.
As far as we know the absolute limitation on TCP packet size is 64K (65535 bytes), and in practicality this is far larger than the size of any packet you will see, because the lower layers (e.g. ethernet) have lower packet sizes. The MTU (Maximum Transmission Unit) for Ethernet, for instance, is 1500 bytes.
I want to know, Is there any any way or any tools, to send packets larger than 64k?
I want to test a device in facing with packet larger than 64k! I mean I want to see, if I send a packet larger than 64K, how it behave? Does it drop some part of it? Or something else.
So :
1- How to send this large packets? What is the proper layer for this?
2- How the receiver behave usually?
The IP packet format has only 16 bit for the size of the packet, so you will not be able to create a packet with a size larger than 64k. See http://en.wikipedia.org/wiki/IPv4#Total_Length. Since TCP uses IP as the lower layer this limit applies here too.
There is no such thing as a TCP packet. TCP data is sent and received in segments, which can be as large as you like up to the limits of the API you're using, as they can be comprised of multiple IP packets. At the receiver TCP is indistinguishable from a byte stream.
NB osi has nothing to do with this, or anything else.
TCP segments are not size-limited. The thing which imposes the limit is that IPv4 and IPv6 packets have 16 bit length fields, so a size larger than this limit is not possible to express.
However, RFC 2675 is a proposed standards for IPv6 which would expand the length field to 32 bits, allowing much larger TCP segments.
See here for a talk about why this change could help improve performance and here for a set of (experimental) patches to Linux to enable this RFC.
What is the behavior of an ethernet device in the packet ingress path?
If the sender is sending a larger-than-MTU frame, then:
1) does the receiver's device drop it directly in hardware,
2) or accept it and send it up for the kernel's IP stack to handle it?
3) when is ICMP frag-required sent?
4) does it make a difference if the ethernet device is say on a intermediate router vs a end host?
It is impossible to answer 1), 2) definitively for all devices and networking stacks. The Ethernet standard defines an MTU of 1500 bytes so that is all you can rely on and in general you should expect frames with larger MTUs to be dropped.
However in reality it is likely in an end host that if the network interface hardware doesn't drop the oversized frame (which is typically referred to as giant) then it will make it's way up the software stack and be processed. Even though the stack may not drop the oversized frame due to it being over the MTU it may still get dropped for other reasons, e.g. due to an internal queue exhaustion.
Although the maximum MTU of an Ethernet frame has remained constant the maximum size of an Ethernet frame has grown over time to encompass features like 802.1Q VLAN single and double tags. MPLS further increases the frame size to include label stacks. This means intermediate switches are usually tolerant of frames that exceed the interface MTU by some amount. One major vendor effectively tolerates a max MTU of 2000 bytes by default in their current switches. Older switches may be less tolerant.
To get a definitive answer you'll need to do some research into the specific hardware and software that you care about.
What is the maximum packet size for a TCP connection or how can I get the maximum packet size?
The absolute limitation on TCP packet size is 64K (65535 bytes), but in practicality this is far larger than the size of any packet you will see, because the lower layers (e.g. ethernet) have lower packet sizes.
The MTU (Maximum Transmission Unit) for Ethernet, for instance, is 1500 bytes. Some types of networks (like Token Ring) have larger MTUs, and some types have smaller MTUs, but the values are fixed for each physical technology.
This is an excellent question and I run in to this a lot at work actually. There are a lot of "technically correct" answers such as 65k and 1500. I've done a lot of work writing network interfaces and using 65k is silly, and 1500 can also get you in to big trouble. My work goes on a lot of different hardware / platforms / routers, and to be honest the place I start is 1400 bytes. If you NEED more than 1400 you can start to inch your way up, you can probably go to 1450 and sometimes to 1480'ish? If you need more than that then of course you need to split in to 2 packets, of which there are several obvious ways of doing..
The problem is that you're talking about creating a data packet and writing it out via TCP, but of course there's header data tacked on and so forth, so you have "baggage" that puts you to 1500 or beyond.. and also a lot of hardware has lower limits.
If you "push it" you can get some really weird things going on. Truncated data, obviously, or dropped data I've seen rarely. Corrupted data also rarely but certainly does happen.
At the application level, the application uses TCP as a stream oriented protocol. TCP in turn has segments and abstracts away the details of working with unreliable IP packets.
TCP deals with segments instead of packets. Each TCP segment has a sequence number which is contained inside a TCP header.
The actual data sent in a TCP segment is variable.
There is a value for getsockopt that is supported on some OS that you can use called TCP_MAXSEG which retrieves the maximum TCP segment size (MSS). It is not supported on all OS though.
I'm not sure exactly what you're trying to do but if you want to reduce the buffer size that's used you could also look into: SO_SNDBUF and SO_RCVBUF.
There're no packets in TCP API.
There're packets in underlying protocols often, like when TCP is done over IP, which you have no interest in, because they have nothing to do with the user except for very delicate performance optimizations which you are probably not interested in (according to the question's formulation).
If you ask what is a maximum number of bytes you can send() in one API call, then this is implementation and settings dependent. You would usually call send() for chunks of up to several kilobytes, and be always ready for the system to refuse to accept it totally or partially, in which case you will have to manually manage splitting into smaller chunks to feed your data into the TCP send() API.
According to http://en.wikipedia.org/wiki/Maximum_segment_size, the default largest size for a IPV4 packet on a network is 536 octets (bytes of size 8 bits). See RFC 879
Generally, this will be dependent on the interface the connection is using. You can probably use an ioctl() to get the MTU, and if it is ethernet, you can usually get the maximum packet size by subtracting the size of the hardware header from that, which is 14 for ethernet with no VLAN.
This is only the case if the MTU is at least that large across the network. TCP may use path MTU discovery to reduce your effective MTU.
The question is, why do you care?
If you are with Linux machines, "ifconfig eth0 mtu 9000 up" is the command to set the MTU for an interface. However, I have to say, big MTU has some downsides if the network transmission is not so stable, and it may use more kernel space memories.
One solution can be to set socket option TCP_MAXSEG (http://linux.die.net/man/7/tcp) to a value that is "safe" with underlying network (e.g. set to 1400 to be safe on ethernet) and then use a large buffer in send system call.
This way there can be less system calls which are expensive.
Kernel will split the data to match MSS.
This way you can avoid truncated data and your application doesn't have to worry about small buffers.
It seems most web sites out on the internet use 1460 bytes for the value of MTU. Sometimes it's 1452 and if you are on a VPN it will drop even more for the IPSec headers.
The default window size varies quite a bit up to a max of 65535 bytes. I use http://tcpcheck.com to look at my own source IP values and to check what other Internet vendors are using.
The packet size for a TCP setting in IP protocol(Ip4). For this field(TL), 16 bits are allocated, accordingly the max size of packet is 65535 bytes: IP protocol details