I am reading about TCP sockets which require a connection to be set up between a client and a server. Is it possible to have TCP sockets working on a wireless connection?
Perhaps you are interpreting 'connection' to mean 'physical link' (as in, wires). This is incorrect.
'physical' in networking extends to all data transfer mediums. In the case of wireless, this is the radio waves. In terms of TCP a 'connection' is made when one computer is listening on a network and responds to a second computer who makes a request.
TCP doesn't care what physical medium it is travelling through.
With regards to your question it may help to think of the OSI model. In this model TCP sits at level 4. What this should mean is that it can be mostly oblivious to what happens above and below it.
The actual physical layer is level 1 of the OSI model, so TCP should not have to care what is happening there. Thus, whether you are using wires, wireless, fiber or pieces of paper to send the TCP packets is irrelevant, so long as the protocol is enacted as described.
Short answer - Yes.
Read more at wiki
I hope so - I am using mobile broadband. And web access uses TCP/Sockets.
The network technology is organized in so called layers.
Physical transmission of signals is responsibility of the first layer. It can be copper cable, optical fiber, radio signals, homing pigeons, whatever technology there is to get zeros and ones between the machines.
TCP is a fourth layer protocol. It relies on the bottom three layers to route packets of data from the source to the destination. It's responsibility is to make sure that the packets arrive in an orderly fashion.
The whole point of this layered structure is that when you program networking drivers, you don't need to worry what kind of data the user will send on the network. When you program web application you don't need to worry if the user is connected with an ethernet CAT-5 cable, or a cable TV modem.
Related
I have some (very) old software written in C, that was used for two devices that communicate via serial cable (RS232) - both sending and receiving messages.
Now the old devices are to be replaced by new modern ones that do not have serial ports, but only Ethernet.
Hence, the request now is to convert the old serial communication to UDP communication (C++ is the choice for the moment).
So, I have some questions about this "conversion":
1) Suppose there are two peers A and B. Should I implement a server and a client for each peer, i.e.: serverA+clientA (for device A) and serverB+clientB (for device B)? Or is there some other/different approach?...
2) The old serial communication had some CRC, probably to ensure some reliability. Is it CRC necessary to be implemented (in my custom messages) also on UDP communication or not?
Thanks in advance for your time and patience.
1) UDP is a connectionless protocol so there's no rigid client and server roles here. You simply have some code that handles receiving and some code that facilitates sending.
2) You don't need CRC for UDP. First, there's a FCS (CRC32) in each Ethernet frame. Then, there's a header checksum in IP packets. After all, checksum is already included in UPD datagram!
Please also consider the following things:
In everyday life COM ports are long gone from the physical world, but they're still with us in the virtual form (even Android phones have COM ports). There are a lot of solutions for doing COM over USB/TCP/whatever. Some of them are PC apps, some of them are implemented in hardware (see Arduino's COM over USB),
When an UDP datagram fails checksum test, it is dropped (usually) silently. So in UDP you don't have built-in capabilities to distinguish between "nothing was received" and "we received something but that's not a valid thing". Check UDP-Lite if you want to handle these situations on the application level (it should simplify the porting process I believe).
Default choice for transferring data is TCP, because it provides reliable delivery. UDP is recommended for users that care about being realtime and for those who can tolerate some data loss. Or for those who care about the resources.
Choose TCP if you are going to send large amount of data or be ready to handle packet congestion on ports. Choose TCP if you plan to go wireless in future or be ready to handle periodical significant loss of packets.
If your devices are really tiny or filled with other stuff, it is possible to operate directly on Level 2 (Ethernet).
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I always hear about the layers of internet and i vaguely understand this. But, what confuse me most is that transport layer (including TCP protocol) lies on top of the internet layer(including IP protocol)..
What does this mean? For one who has a foggy understanding of the internet mechanism (I'm not a CS student or something I am just a hobby programmer)
The picture I have about the internet is that the network card sends/receives signals (packets) from the internet through wired connection / wifi then the OS using socket API sends/receives these packets acting as a layer between the hardware and the application which in turn uses some high-level protocol such as HTTP to interpret the data transferred - these protocol usually provided by languages e.g. python or java
.. I guess then that IP and TCP protocol are used at the level of the socket API? but I need more details ? I hope the explanation be in terms of coding/programming/implementation because abstractions used in this area confuse me.
Thank you , and sorry for my bad English
This is part of a layered solution to solve networking. Each layer has its own functionality:
IP (Internet Protocol) is in charge of delivering a packet (or datagram) from one interface, in one machine, with an IP address assigned to another interface in the same or other machine (node). Both nodes can be in the same LAN or different LAN connected through different paths (LAN's and routers). Basically it will make the packet get from source IP to destination IP. It provides a best-effort services, it doesn't assure the IP packet is going to arrive, it can be lost in the middle.
Above layer 3 or IP in the so-called TCP/IP stack, there is the transport layer. Its main functionality is to multiplex the lower layer (IP) service (take a packet from src to dst) among different applications. This is why in all transport layer protocols there is the concept of port or more generically Transport Service Access Point (TSAP). UDP, TCP, SCTP do that. UDP provides an unreliable service to the application. TCP provides a connected, reliable transport service to the application. This layer will make a message sent from application A in node Y reach application A1 in node Z, either reliably or unreliably (while IP only takes care of carrying the packet from node Y to node Z).
You will need to read a little about the OSI layered model and the TCP/IP layered model.
If you need to get more info I can address you to a training I have about IPv6 with a good introduction to networking: http://www.slideshare.net/rodolk/networking-tcpip-stack-introduction-ipv6
TCP is a protocol, known as "Transmission Control Protocol" - by specification it has features in place which makes sure that transmitted data is checked. On the other hand, there are things such as UDP, aka "User Datagram Protocol" which also works on top of IP - by specification this method does not check any transmitted data, so it's less useful where files must be fully intact (more utilised for video streaming, where some lost frames is acceptable, as opposed to binary file transfers where incorrect data means corruption and the whole file would be useless).
On to IP, IP is an addressing protocol, allowing a network to address and communicate with any machine that lives within it. IP stands for Internet Protocol, and it defines the fundamental way that two machines communicate over the "internet". It does not define how communications are handled, in ways such as being checked for data integrity, etc.
So, to summarise, the TCP and UDP are just extensions of IP. It is entirely possible, however, to have a socket based TCP or UDP connection, and I expect it's also possible to have some sort of MAC address protocol (as opposed to an IP address protocol). I don't know of any protocols which are similar to IP, but I imagine they do exist. In reality, using TCP over something other than IP is entirely unlikely. If you're going to the effort to create a custom protocol, chances are you'll want it fully custom and won't want to stick to design specifications designed for another protocol layer.
Note that calling it a "TCP/IP" connection is probably only ever used for legacy reasons. A lot of terms like this still exist because before the technology "bubble" growth, there were competing alternatives to IP. Even today, there is IPv6 which is technically an alternative to IPv4. It's also possible that we might one day outgrow IPv6, and at that point in time, there could be something other than IP to worry about.
I'm about to re-architect a real-time system that has been prototyped on a single node and specify how it should be scaled up to multiple nodes (probably never more than 20 of them in any one LAN). Some of the functionality will multiply on a per-node basis, and some of it will remain centralised on a one-per-system basis. There is going to be a need for communication between each node and that central unit (possibly a master node), but not between individual nodes.
Due to the real-time demands of the system, UDP is something that should be considered for that communication. But... it is almost always described as unreliable. Is this always the case? Does it not depend on the scale of the network, the data load on the network and the way the protocol is used?
For example, suppose I have a central unit which regularly polls through each node by addressing a UDP message to it, and each node immediately responds with its data via UDP. There is no other communication on the (isolated) network. Suppose there is also some mechanism to ensure there are never any collisions (e.g. all nodes have a maximum transmission length for their responses to a poll message, and the latencies are nailed down to known levels). Is there any (hidden) reason in a simple and structured network like this that you would ever fail to transmit/receive every last UDP packet and have near 100% reliability?
EDIT: the detail of this question suffers from confusion around what "unreliable" means, and whether it is intended to apply only to UDP, or to the system in which UDP is employed. I have chosen to leave this confusion in the question, because looking back over a lot of material on UDP, I can see that this confusion might be very common, and that answers which highlight that confusion and overcome it might be valuable.
The key is, UDP does not make any guarantees. There are many reasons why datagrams might go undelivered:
Sender host buffers fill up
Cosmic rays flip bits somewhere along the way, causing a checksum mismatch and the datagram to be discarded
Electromagnetic interference corrupts the signal momentarily
A network cable gets unplugged for a moment
A hub or switch loses power for a moment
A switch's buffers fill up
Receiving host buffers fill up
If any of these things (or many others) occurs, a datagram may go undelivered. UDP will make no attempt to detect this or to re-deliver it.
Yes. Every layer is potentially unreliable, starting with the electrical signalling across your Ethernet cable. (Ever jostled one of those plugs? You can see it in Wireshark logs.) Collisions are virtually impossible to avoid. And in case of congestion, your protocol stack may decide to drop UDP packets.
But all that's rather beside the point. UDP is unreliable, but that doesn't mean it can't be relied on. Plenty of mission-critical applications run over UDP. You just need to understand the unreliability and account for it.
Unreliable does not mean it will definitely fail. It only means that it does not care about transport problems and thus will not make any guarantees that transmission will be successful. Let's compare some aspects of UDP against TCP.
UDP is packet based, TCP stream based. This has not much to do with reliability.
Packets may arrive in a different order than they were sent. UDP does not care and will deliver the packets in this order to the application. In TCP data have a sequence number so the receivers operating system will detect reordering and forward the data to the application in the correct order. This usually does not matter when you have a direct connection between client and server, but might happen in wide networks like the internet.
Packets may get lost due to router or switch congestion or overload of the senders or receiving system or others. This might also happen in local networks with heavy traffic or if the receiver system is unable to cope with the amount of data, even for a short time. With UDP the data will be lost. TCP instead will detect lost packets and retransmit them and even slow down the traffic to adapt to what speed network and endpoints can handle and thus loose less packets in the future.
Packets might get duplicated. Again TCP will detect this due to the sequence number but UDP will not and thus transmit the duplicate packet to the application.
Packets might get corrupted. Both TCP and UDP have the same kind of checksum to detect small errors, but will not detect larger errors.
Applications using UDP usually does not need the reliability of TCP or don't need all of this. For instance with real time audio and video packet loss is acceptable but duplicates and reordering is not. Thus the RTP protocol contains its own sequence number (timestamp) to detect this case. Also, RTP is often accompanied by the RTCP protocol to send statistics about packet loss back to the peer and thus make adaption of connection speed possible.
If you want reliable UDP, try looking at ENet library.
http://enet.bespin.org/
Unreliability with regard to UDP is different from unreliability in general. Also, UDP and alternatives to it (e.g. TCP) are always only ever components or single layers in a wider system. This can lead to some confusion about what "unreliable" means.
UDP is a transport layer network protocol. The transport layer is responsible for getting data from one point on the network to another specific point on the network. In that context, UDP is described as an "unreliable" protocol because it makes no guarantees about whether the data sent will actually arrive. In contrast, TCP is a "reliable" transport layer protocol because if data goes missing or is corrupted the first time it is sent, the protocol itself has mechanisms to resend the data and ensure it arrives... eventually.
But UDP is not some sloppy "maybe, maybe not - let me think about it and screw you around" protocol. It does what it is specified to do, and is reliable (general sense) at doing it... as well as reliable (general sense) in failing in predictable ways. If you take these failure modes into account elsewhere, UDP can be a component of an overall very reliable system.
For example, by restricting network topology and using UDP to transport higher level protocols, the GigE Vision standard specifies a highly reliable system with high data transfer rates and real-time response whose transport level communications is dominated by UDP traffic.
Historically, the major source of unreliable packet transport was packet collisions due to two sources attempting to transmit simultaneously on a single channel. In modern networks, each node is typically connected on a full duplex link to a network switch, making collisions impossible on that link, and consequently making modern networks much more reliable (in all senses) than was the case when UDP was first designed.
No networking technology currently available can be made 100% reliable... but let's be practical rather than pedantic, because potential unreliability and actual unreliability are a lot like shark attacks - they tend to occur far more in people's minds than in reality.
Some material on UDP makes it sound almost like the people who designed UDP did it just to annoy people - that unreliability was deliberately engineered in. This is not the case, and it is unhelpful to think of it in these terms. It is far better to focus on what UDP does and does not do in comparison to alternatives (e.g. see this comparison between TCP and UDP... which nonetheless lists "unreliability" as a key feature of UDP).
In reality, when there is data to be transmitted, that can be transmitted, it is transmitted; when there is data that can be received, it is received. Likewise, if you transmit packets 1, 2 then 3 directly to an endpoint, they will almost certainly be received as packets 1, 2 and 3 in order (assuming no failures in lower network layers, and that incoming data is buffered in a FIFO as is customary, but not mandatory). You can get a lot of reliability out of this, depending on how you use it.
However, if you transmit packets via multiple routes, all bets are off - "unreliability" of packet order can occur. And if you flood the available buffers, unreliability via dropping packets will occur. And if you allow nodes to transmit at any time (asynchronous), then you will get unreliability through packet collisions. But in the "simple and structured" (and also small and synchronous) LAN described, you may be able to either avoid this, or detect its occurrence (e.g. by sending an incrementing counter value in each packet), which will let you compensate in an application-specific way.
In cases where the power goes off (perhaps momentarily), or cosmic rays strike, or people trip on loose cables causing an unacceptable level of "unreliability"... then don't blame UDP - blame the engineer(s) whose design left the system susceptible to these things.
All things considered, in the LAN described, you might reasonably expect to be able to engineer a system based on UDP so as to never lose more than one packet in every few million, or billion, or even astronomically better than this - but it will depend on specifics, and only you can know if your application can tolerate the quantity and quality of unreliable comms that results in your case.
I have no experience with sockets nor multiplayer programming.
I need to code a multiplayer mode for a game I made in c++. It's a puzzle game but the game mode will not be turn-based, it's more like cooperative.
I decided to use UDP, so I've read some tutorials, and all the samples I find decribes how to create a client that sends data and a server that receives it.
My game will be played by two players, and both will send and receive data to/from the other.
Do I need to code a client and a server?
Should I use the same socket to send and receive?
Should I send and receive data in the same port?
Thanks, I'm kind of lost.
Read how the masters did it:
http://www.bluesnews.com/abrash/chap70.shtml
Read the code:
git clone git://quake.git.sourceforge.net/gitroot/quake/quake
Open one UDP socket and use sendto and recvfrom. The following file contains the functions for the network client.
quake/libs/net/nc/net_udp.c
UDP_OpenSocket calls socket (PF_INET, SOCK_DGRAM, IPPROTO_UDP)
NET_SendPacket calls sendto
NET_GetPacket calls recvfrom
Do I need to code a client and a server?
It depends. For a two player game, with both computers on the same LAN, or both on the open Internet, you could simply have the two computers send packets to each other directly.
On the other hand, if you want your game to work across the Internet, when one or both players are behind a NAT and/or firewall, then you have the problem that the NAT and/or firewall will probably filter out the other player's incoming UDP packets, unless the local player goes to the trouble of setting up port-forwarding in their firewall... something that many users are not willing (or able) to do. In that case, you might be better off running a public server that both clients can connect to, which forwards data from one client to another. (You might also consider using TCP instead of UDP in that case, at least as a fallback, since TCP streams are in general likely to have fewer issues with firewalls than UDP packets)
Should I use the same socket to send and receive?
Should I send and receive data in the same port?
You don't have to, but you might as well -- there's no downside to using just a single socket and a single port, and it will simplify your code a bit.
Note that this answer is all about using UDP sockets. If you change your mind to use TCP sockets, it will almost all be irrelevant.
Do I need to code a client and a server?
Since you've chosen to to use UDP (a fair choice if your data isn't really important and benefits more from lower latency than reliable communication), you don't have much of a choice here: a "server" is a piece of code for receiving packets from the network, and your "client" is for sending packets into the network. UDP doesn't provide any mechanism for the server to communicate to the client (unlike TCP which establishes a 2 way socket). In this case, if you want to have two way communication between your two hosts, they'll each need server and client code.
Now, you could choose to use UDP broadcasts, where both clients listen and send on the broadcast address (usually 192.168.1.255 for home networks, but it can be anything and is configurable). This is slightly more complex to code for, but it would eliminate the need for client/server configuration and may be seen as more plug 'n play for your users. However, note that this will not work over the Internet.
Alternatively, you can create a hybrid method where hosts are discovered by broadcasting and listening for broadcasts, but then once the hosts are chosen you use host to host unicast sockets. You could provide fallback to manually specify network settings (remote host/port for each) so that it can work over the Internet.
Finally, you could provide a true "server" role that all clients connect to. The server would then know which clients connected to it and would in turn try to connect back to them. This is a server at a higher level, not at the socket level. Both hosts still need to have packet sending (client) and receiving (server) code.
Should I use the same socket to send and receive?
Well, since you're using UDP, you don't really have a choice. UDP doesn't establish any kind of persistent connection that they can communicate back and forth over. See the above point for more details.
Should I send and receive data in the same port?
In light of the above question, your question may be better phrased "should each host listen on the same port?". I think that would certainly make your coding easier, but it doesn't have to. If you don't and you opt for the 3rd option of the first point, you'll need a "connect back to me on this port" datafield in the "client's" first message to the server.
I've been reading up on basic network programming, but am having a difficult time finding a straight-forward explanation for what exactly and socket is, and how it relates to either the OSI or TCP/IP stack.
Can someone explain to me what a socket is? Is it a programmer- or API-defined data structure, or is it a hardware device on a network card?
What layers of the mentioned network models deal with "raw" sockets? Transport layer? Network layer?
In terms of the data they pass between them, are socket text-based or binary?
Is there an alternative to sockets-based network programming? Or do all networked applications use some form of socket?
If I can get this much I should have a pretty clear understanding of everything else I'm reading. Thanks for any help!
Short answers:
Socket is an abstraction of an IP connection endpoint - so if you think of it as an API structure, you are not very far off. Please do read http://en.wikipedia.org/wiki/Internet_socket
Internet layer i.e. IP Protocol. In practice you usually use explicitly sockets that bind to a certain transport layer parameters (datagram/UDP or stream/TCP)
Sockets send data, in network byte order - whether it is text or binary, depends on the upper layer protocol.
Theoretically, probably yes - but in practice all IP traffic is done using 'sockets'
Socket is a software mechanism provided by the operating system. Like its name implies, you can think of it like an "electrical outlet" or some electrical connector, even though socket is not a physical device, but a software mechanism. In real world when you have two electrical connectors, you can connect them with a wire. In the same way in network programming you can create one socket on one computer and another socket on another computer and then connect those sockets. And when you write data to one of them, you receive it on the other one. There are also a few different kinds of sockets. For example if you are programming a server software, you want to have a listening socket which never sends or receives actual data but only listens for and accepts incoming connections and creates a new socket for each new connection.
A socket, in C parlance, is a data structure in kernel space, corresponding to one end-point of a UDP or TCP session (I am using session very loosely when talking about UDP). It's normally associated with one single port number on the local side and seldom more than one "well-known" number on either side of the session.
A "raw socket" is an end-point on, more or less, the physical transport. They're seldom used in applications programming, but sometimes used for various diagnostic things (traceroute, ping, possibly others) and may required elevated privileges to open.
Sockets are, in their nature, a binary octet-transport. It is not uncommon to treat sockets (TCP sockets, at least) as being text-based streams.
I have not yet seen a programming model that doesn't involve something like sockets, if you dig deep enough, but there have certainly been other models of doing networking. The "/net/" pseudo-filesystem, where opening "/net/127.0.0.0.1/tcp/80" (or "tcp/www") would give you a file handle where writes end up on a web server on localhost is but one.
Suppose your PC at home, and you have two browser windows open.
One looking at the facebook website, and the other at the Yahoo website.
The connection to facebook would be:
Your PC – IP1+port 30200 ——– facebook IP2 +port 80 (standard port)
The combination IP1+30200 = the socket on the client computer and IP2 + port 80 = destination socket on the facebook server.
The connection to Yahoo would be:
your PC – IP1+port 60401 ——–Yahoo IP3 +port 80 (standard port)
The combination IP1+60401 = the socket on the client computer andIP3 + port 80 = destination socket on the Yahoo server.