the version field in the IP header contains the version of the IP protocol used (IPv4/IPv6).
the type field in the Ethernet header contains which layer 3 protocol to forward the De-encapsulated packet to (IPv4, IPv6).
Why is it so important that the IP header have a field called "Version", what is it used for?
Networking protocols are designed in layered manner to reduce complexities and improve generalities. The IP layer is designed in such a way that each of system can have a suitable IP layer software to process the packets without worrying about how the packets are transferred. This way IP packets can be transported by ethernet layer or any other layer (e.g. FC or ATM) independent of the knowledge of layer below it. The IP processing software needs to know the version of IP and hence IP packets have that version.
The same layering concept applies to ethernet layer which can help in transporting not only IP packets but other type of packets (e.g. IPX) without knowing details about protocol that it is transporting. But the ethernet layer processing at each end should know what type of packets they are carrying, so that they can hand-over the packet to appropriate layer for further processing.
Because IP can be encapsulated in another (non-Ethernet) layer 2 protocol. The IP protocol does not mandate a specific lower layer protocol so it can not make assumption on information that is available at this layer (e.g. the upper layer protocol version number). So if the layer 2 protocol only specifies 'IP' as the next layer, you need the IP protocol version from the IP header to correctly process it.
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As I understand it TCP is required for congestion control and error recovery or reliable delivery of information from one node to another and its not the fastest of protocols for delivering information.
Some routing protocols such as EIGRP and OSPF ride directly on top of IP. Even ICMP rides directly over IP.
Why is UDP even required at all? Is it only required so that developers/programmers can identify what application the inbound packet should be sent to based on the destination port number contained within the packet?
If that is the case then how is information gathered from protocols that ride directly on top of IP sent to the appropriate process when there is no port number information present?
Why are voice and video sent over UDP? Why not directly over IP?
(Note that I do understand thoroughly the use case for TCP. I am not asking why use UDP over TCP or vice versa. I am asking why use UDP at all and how can some protocols use directly the IP layer. Whats the added advantage or purpose of UDP over IP?)
Your question makes more sense in terms of why is UDP useful (than why is UDP required).
UDP is a recognized protocol by the Internet Assigned Numbers Authority. UDP can be useful if you want to write a network protocol that's datagram based and you want to play more nicely with Internet devices.
Routers can have rules to do things like drop any packet that doesn't make sense. So if you try and send packets using say an unassigned IP protocol number between hosts separated by one or more routers, the packets may well never get delivered as you've intended. The same could happen with packets from an unrecognized UDP protocol but that's at least one less door to worry about whether your packet can make it through.
Internet endpoints (like hosts) may do similar filtering too. If you want to write your own datagram based protocol and use a typical host operating system, you're more likely to need to write your software as a privileged process if not as a kernel extension if you're trying to ride it as its own IP protocol (than if you'll be using UDP).
Hope this answer is useful!
First of all, IP and UDP are protocols on the different layers, IP by definition is Internet layer when UDP is transport layer. Layers were introduced to simplify network protocols architecture and to separate concerns. Application layer protocols are supposed to be based on transport layer (with some exceptions).
Most popular transport protocols (in IP network) are UDP and TCP. While TCP is feature rich but with many tradeoffs UDP is very simple but gives a lot of freedom and so typically is a base for other protocols.
The main feature of UDP is multiplexing: ports that allow multiple protocol instances (aka sockets) to coexist on the same node. This means that implementing your own protocol over IP instead of UDP either you won't be able to have multiple instances of your protocol on the same machine or you'll have to implement multiplexing yourself.
There're other features like segmentation and checksum. These features are not mandatory.
And as was mentioned in another answer there're lots of middleware like routers, NATs and firewalls that can ruin the idea of a custom "right over IP" protocol, but it's more like a collateral damage than a feature of UDP.
I think at L2 it will be ether type and destination MAC address.
at L3 it can be protocol field and Destination IP Address.
I want to know what are the other fields by which we can differentiate between control and data packet.
#Anupam Thakur: In pure L2 infrastructure, on receiving frame, node does the bridge-table/l2-table/cam lookup based on destination mac address. For different BPDUs (almost always reserved multicast mac dst addresses, few exceptions beyond the scope of this question) wherever required cam table is programmed with punt path to control-plane module (/also). Note, punt path doesn't distinguish between different protocols. It's job is to just push the packet in Q towards control-plane module. When cp module receives such packet, based on ethertype or LLC content it determines the further protocol processing of the packet.
"Router Alert Option" in IP header can also force packet to be punted to Control Plane, even though it is not destined for recipient device. (i.e. transit node)
packet switching is a protocol, where a message received from tcp layer is divided into packets only at sender machine ip layer and each packet is sent individually on different routes with an identification field set in ip header to help use re-assemble at destination machine.
Where as
fragmentation at ip layer is done on sender machine or any of the, on the way layer 3 device ip layer and fragmentation field is set in ip header to help use re-assemble at destination machine only.
My question:
is my understanding correct?
In packet switching, If message could not be re-assembled due to missing packet at destination, based on identification field, that message is discarded at ip layer of destination machine and tcp layer of sender machine will take care of retransmission of that message, am i correct?
packet switching is a protocol
No. Packet switching is an alternative to circuit switching at the physical layer.
where a message received from tcp layer is divided into packets only at sender machine ip layer and each packet is sent individually on different routes with an identification field set in ip header to help use re-assemble at destination machine.
None of this is correct as a description of packet switching. Packet switching implies the existence of packets, period. It doesn't impose any of these constraints.
Whereas fragmentation at ip layer is done on sender machine or any of the
... intermediate nodes
on the way layer 3 device ip layer
Layer 2
and fragmentation field is set in ip header to help use re-assemble at destination machine only.
My question:
is my understanding correct?
No. You seem to think that packet switching and fragmentation are in some kind of opposition. They aren't. Fragmentation is an extension of packet switching if anything, not an alternative to it.
In packet switching, If message could not be re-assembled due to missing packet at destination, based on identification field, that message is discarded at ip layer of destination machine and tcp layer of sender machine will take care of retransmission of that message, am i correct?
No. Again you're confused about what packet switching is. Your remark applies pretty well to TCP, but that's because of the semantics of TCP, not because of packet switching.
Out of curiosity can we implement a TCP stack directly over Ethernet and use that for reliability and flow control on a LAN network.
TCP in tightly integrated on IP. One example is its use of a pseudoheader which incorporates specific parts of the IP header for purposes of checksum calculation. Also, I doubt if any TCP implementations exist which don't contain pervasive assumptions of 32-bit (for IPv4) or 128-bit (for IPv6) addresses.
Why would you want to skip IP and run TCP directly over a lower layer? Just to save 20 bytes of header?
Why is the Pseudo header prepended to the UDP datagram for the computation of the UDP checksum? What's the rational behind this?
The nearest you will get to an answer "straight from the horse's mouth", is from David P. Reed at the following link.
http://www.postel.org/pipermail/end2end-interest/2005-February/004616.html
The short version of the answer is, "the pseudo header exists for historical reasons".
Originally, TCP/IP was a single monolithic protocol (called just TCP). When they decided to split it up into TCP and IP (and others), they didn't separate the two all that cleanly: the IP addresses were still thought of as part of TCP, but they were just "inherited" from the IP layer rather than repeated in the TCP header. The reason why the TCP checksum operates over parts of the IP header (including the IP addresses) is because they intended to use cryptography to encrypt and authenticate the TCP payload, and they wanted the IP addresses and other TCP parameters in the pseudo header to be protected by the authentication code. That would make it infeasible for a man in the middle to tamper with the IP source and destination addresses: intermediate routers wouldn't notice the tampering, but the TCP end-point would when it attempted to verify the signature.
For various reasons, none of that grand cryptographic plan came to pass, but the TCP checksum which took its place still operates over the pseudo header as though it were a useful thing to do. Yes, it gives you a teensy bit of extra protection against random errors, but that's not why it exists. Frankly, we'd be better off without it: the coupling between TCP and IP means that you have to redefine TCP when you change IP. Thus, the definition of IPv6 includes a new definition for the TCP and UDP pseudo header (see RFC 2460, s8.1). Why the IPv6 designers chose to perpetuate this coupling rather than take the chance to abolish it is beyond me.
From the TCP or UDP point of view, the packet does not contain IP addresses. (IP being the layer beneath them.)
Thus, to do a proper checksum, a "pseudo header" is included. It's "pseudo", because it is not actaully part of the UDP datagram. It contains the most important parts of the IP header, that is, source and destination address, protocol number and data length.
This is to ensure that the UDP checksum takes into account these fields.
When these protocols were being designed, a serious concern of theirs was a host receiving a packet thinking it was theirs when it was not. If a few bits were flipped in the IP header during transit and a packet changed course (but the IP checksum was still correct), the TCP/UDP stack of the redirected receiver can still know to reject the packet.
Though the pseudo-header broke the separation of layers idiom, it was deemed acceptable for the increased reliability.
"The purpose of using a pseudo-header is to verify that the UDP
datagram has reached its correct destination. The key to
understanding the pseudo-header lies in realizing that the correct
destination consists of a specific machine and a specific protocol
port within that machine. The UDP header itself specifies only the
protocol port number. Thus, to verify the destination, UDP on the
sending machine computes a checksum that covers the destination IP
address as well as the UDP datagram. The pseudo-header is not
transmitted with the UDP datagram, nor is it included in the length."
E. Comer - Internetworking with TCP/IP 4th edition.
Pseudo IP header contains the source IP, destination IP, protocol and Total length fields. Now, by including these fields in TCP checksum, we are verifying the checksum for these fields both at Network layer and Transport layer, thus doing a double check to ensure that the data is delivered to the correct host.