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I would like to create a new protocol, i.e having features of TCP and UDP. can you tell me what are tips and techniques are required if it is possible.
Thanks in advance
TCP gives you three features that UDP does not: (a) estimating sending rate, (b) retransmission, and (c) flow-control. In doing so, the tradefoff is that TCP becomes slower compared to UDP. So, if your application is delay sensitive, which is typically true for audio/video applications, then you need to start with UDP and keep whichever of the above three you want. Typically, UDP applications might add forward-error-correction or application layer packet-book-keeping to ensure retransmission.
There is yet another advantage that UDP offers which TCP does not: if you have an application that might use mulitcast. For such cases, UDP would be the right chioce since UDP can handle point to multipoint. Using TCP for multicast applications would be hard since now the sender would have to keep track of retransmissions/sending rate for multiple receivers.
So, in summary, UDP will offer you two features that TCP cannot: lower-delay and ability to do multicast. So, this way, we can actually reduce the scope of the question and ask what are the features of TCP that one would like to add to UDP since there is no way, one can add features of UDP to TCP.
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Before a browser can request for a webpage, A TCP connection is required to be established. What is the necessity of 3 way handshake in interacting with a server computer? why can't we simply send a web request and wait for the response?
Shouldn't the resolution of IP address be enough for this purpose?
Basically, I need to know the reason for establishing TCP connection.
thanks in advance
You are using a device named A and server is named B
Host A sends a TCP SYNchronize packet to Host B
Host B receives A's SYN
Host B sends a SYNchronize-ACKnowledgement
Host A receives B's SYN-ACK
Host A sends ACKnowledge
Host B receives ACK.
TCP socket connection is ESTABLISHED.
See more at: http://www.inetdaemon.com/tutorials/internet/tcp/3-way_handshake.shtml#sthash.F2f4b8Xn.dpuf
Because you need a TCP connection to send HTTP over, and TCP has a 3-way handshake.
Basically, I need to know the reason for establishing TCP connection.
Because HTTP runs over TCP. It doesn't exist in a vacuum.
TCP provides ordering, automatic retransmission, and congestion control. I'd say these are the obvious reasons why the design adopted TCP.
In contrast, e.g. UDP is fast. There is no handshaking. But UDP packets are not ordered, also packets can get lost (no automatic retransmission), and there is no congestion control.
You can try implementing your data transferring for things like HTML in UDP. It's not easy, you still need to reinvent ordering and retransmission for reliable lossless delivery.
If you don't care about lossy or a bit out-of-order transferring then you probably don't need TCP. (e.g. real time video)
--
On the other hand, avoid TCP to get better performance is not necessarily a bad idea. Read about QUIC. (It also has features like loss recovery and congestion control, you should not expect it to be extremely lightweight.)
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Is it possible to convert all outgoing udp traffic from a gateway router to tcp on the same port with iptables. Have looked at mangle but unsure how I can use that.
This is not possible in a generic way for at least the following reasons:
UDP is a datagram protocol, where each packet is independent from the other. TCP instead is a stream protocol. While it would be possible to concatenate the UDP packets to a TCP stream it is not clear at which boundaries the TCP stream should be split to generate the packets.
With UDP duplicates, packet loss and reordering of packets could happen. If you cannot be sure about the proper order and that you got all messages and exactly once you can not construct reliable a TCP stream from it.
TCP acknowledges when it receives packets and if an ACK is missing the packet gets retransmitted. Once you forward TCP to UDP these ACKs can not be done because UDP does not have such mechanism.
It might still be possible to do application specific translations, like UDP DNS to TCP DNS or similar. But this translation depends on the application protocol so you need to have an application specific gateway. iptables as a packet filter works on a lower layer and does not provide such functioniality.
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to settle an argument at work.
Is it possible, in theory, to create a network protocol with ports higher than 65535 and get it approved by IANA ?
if not, why not ?
Yes. It is possible to create one with an arbitrary number of ports.
Getting it approved by IANA (and IETF, ISO and whoever else are concerned) is just a matter of getting enough people to use your wonderful new invention. Which is extremely unlikely but not theoretically impossible.
If you are using TCP or UDP on top of IP, then no. Both TCP and UDP only use 16 bits for the port number. If you are using some other protocol suite such as IPX/SPX, then it depends on that protocol. If you are making your own protocol, then you can do whatever you want. Getting it approved by IANA is a different story.
http://en.wikipedia.org/wiki/Transmission_Control_Protocol
http://en.wikipedia.org/wiki/User_Datagram_Protocol
2^16 = 65536 different ports
Since computer numbers start counting at 0, this leaves 65535 as the max.
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I am trying to understand TCP congestion avoidance mechanism, but I don't understand one thing:
TCP congestion avoidance is per flow or per link?
In other words: there are 2 routers A and B
A is sending B two TCP flows - when one TCP flow detects congestion, does it decrease window size in the other flow as well?
of course if this happens, the other flow will detect congestion in some time, but does the second flow "waits" until it detects congestion on its own? that would be quite uneffective...
thanks a lot
It decreases the window size for the current connection. Each connection's RTT and windows are maintained independently.
Routers operate on layer 3 (IP) and are not aware of layer 4 (TCP), because of this, routers do not take any part in TCP congestion avoidance mechanism. This mechanism is fully implemented by TCP endpoints. It is triggered by routers dropping IP packets, but (classic) routers are not aware what higher level protocol IP packets carry.
The fact that one flow does not affect the other is quite desirable from the security perspective. With NAT you can have many hosts sharing the same IP address. From the outside world all these hosts look as a single machine. So, if some server reduced throughput of all TCP connections coming from a single IP address in response to packets dropped within one of those connections that would open a door to quite nasty DoS attacks.
Another issue is that some routers may be configured to drop packets based on IP ToS field. For example, latency sensitive SSH traffic may set different ToS than bulk FTP download. If router is configured to take into account ToS field, it may drop packets belonging to FTP connection, which should trigger congestion avoidance, but is should not affect packets belonging to SSH connection, which may be handled with higher priority.
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I want to connect two computer via serial but for each to see each other via a TCP/IP connection. Ie, create new network device on the computers that are in actual fact serial ports.
The reason for this is that I am actually testing the medium in which the serial connection is made (wireless), and part of the experiment will be to use TCP/IP.
The radio being tested is a telemetry radio for use in low power applications. It polls once a second, sending data out on the wireless channel every poll when something has been received via the serial port. It uses a Modbus RTU delimiter to determine the end of data coming in on the serial port.
slip and ppp are more suitable for use with actual serial modems from what I understand.
This is actually a very hard problem. TCP/IP is a very chatty protocol and you will have problems with the radio system you have described because of the pattern of packets and ACKs you will have. In the past for some similarly unsuited applications I worked on a system that fibbed about the TCP/IP connection by faking some packets while pushing the data over a link like you have.
It is a pain, but we were doing it to support sshing over a totally inappropriate channel (high loss and high latency with moving endpoints) but it worked.
SLIP (Serial Line IP) sounds like something you might want to look into for this project.
you may use Simple TCP/IP, UDP connection by using UART using a software like this:
http://www.serialporttool.com/CommTunnel.htm