I have several apps which communicate through serial comm (RS-232 and RS-422), and i would like them to communicate through TCP or UDP without changing them. Another point is that some of the apps must run on linux.
I would like to know if there are exsiting tools for that purpose..
Thanks a lot!
Tal
If all you do with your serial port is read and write bytes, and if precise timing is not a concern, then you may be able to replace your serial port object with a TCP socket and send the exact same data over the socket as you would have sent over the port. The biggest complications are that the timing on a TCP socket is much looser than on a serial port, and TCP sockets' mechanisms for sending "out-of-band" data are different from those of a serial port.
I am unaware of any standards for sending serial data via UDP. Conceptually, it would seem like a useful thing to have since there are many serial-port protocols-based in which it would be more useful to drop data that can't be delivered within a certain time frame than to deliver it late. For example, if the intended recipient of serial-port data is an embedded controller that will sometimes so busy that it drops some incoming data, but which will respond within a few milliseconds to everything it does receive, a one-second hiccup on a TCP connection [not unusual] may cause software which expects to be talking directly to the controller to retransmit a command a dozen times. Even if the device would be capable of detecting and rejecting retransmissions it receives, it would be better for the earlier transmission requests to be abandoned than to have them be delivered late. Note that to be useful, a UDP-based scheme would have to include enough wrapper logic to guarantee that packets would never be delivered out of order; once the data for a packet has been sent to the serial port, any UDP packets which are received later despite having been sent earlier must be discarded; the recipient should probably include logic so that if many out-of-sequence packets are received, it will wait a little while after receiving any packet whose sequence number does not immediately follow the last one, to see if missing packets show up before it commits itself to discarding them.
There is no standard tool to do that. I am thinking to develop one. UDP is ideal in this case since it is 100% guaranteed that there is no out of order packet delivery on a short LAN as is in your case.
In several projects I have used free tool Hercules (https://www.hw-group.com/software/hercules-setup-utility) for protoyping and testing phases. No advertising intended, just a recommendation.
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).
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 an application that communicates in real time with other clients over LAN. The application requires packets to be in order and all to arrive. It also requires as fast transfer as possible and I seem to have some problems with TCP in this matter.
So I was thinking about this, as a non-experienced network programmer, what if I first send a UDP protocoled message and then the same data with TCP. If the UDP-message arrive I will have it as fast as possible if not I still have the TCP message that will make sure I'll atleast get the packet. Obviously I'll make sure that I don't read the same data twice by giving each message an ID or similar.
Is this any good approach? I was thinking that maybe sending the tcpmessage simultaneously will just slow the udp message down so It wont make a difference anyways.
No, this is not a good approach.
You are doubling your network bandwidth and significantly increasing the complexity of your networking code for very little gain.
TCP and UDP have very different characeristics. If you care about data arriving in a timely manner, where if data is late it is no use, then TCP is not useful and as such you should use and only use UDP. If you do not care about data arriving in a timely manner, then UDP is not useful, as it is not reliable.
UDP has very specific use cases. i.e. say an online game which sends a players co-ordinates. You state the order and acknowledgment is needed, therefore TCP seems like the most sensible approach.
Although just to put a twist in the mix, TCP can sometimes surprise you and be better performance wise under specific circumstances.
TCP will try and buffer the data before it sends it across the network (more efficient use of bandwidth). UDP on the other hand puts a packet across the network immediately.
Now imagine writing lots of small packets across a network, UDP may cause congestion whereas TCP is better controlled.
No it is not a good approach at all. You will have now twice the data being sent.
For real time communication UDP is the best approach. You should design your reciever algorithm to manage out of data arrival and sort it and also non arrival of some data.
Also the kind of data being sent can be a deciding factor. If its transactions of a financial kind, udp is not a good idea. But then you should be on a different network.
If it is video data, real time is very important, losses can be tolerated.
So see if you can use the properties of your data to manage udp connection well.
While sending packets across a network, how can one determine where one packet ends and where another starts?
Is sending/receiving acknowledgment one of the ways of doing so?
TCP is a stream-based protocol. That is, it provides a stream vs. packet or message-based interface to the application. If using TCP, an application must implement its own method of determining packets or messages. For example, (a) all message are a fixed size, or (b) each message is prefixed with its subsequent size, or (c) there is a special "end-of-record" sequence in the data stream to indicate a message boundary. Search google for lots of information on how one can implement message boundaries in TCP.
I assume here that you mean application-level 'packets'.
If you use UDP, you don't need to since it's a message protocol. TCP is a byte streaming protocol, so it cannot send packets, just bytes. If you need to send anything more complex than a byte-stream across TCP, you have to add another protocol on top - HTTP is one such protocol. Text is fairly easy since lines have terminating characters, usually CR/LF/CRLF. Sending non-text messages will require a different protocol.
One approach that is often used with TCP is to connect, stream a protocol-unit, disconnect. This works OK, but slowly because of the huge latency of continually opening and closing TCP connections. HTTP usually works like this in order to serve up web pages to large numbers of users who, if left permanently connected while they viewed pages, would needlessly use up all the server sockets.
Waiting for an application-level ACK from the peer is sometimes necessary if it absolutely essential that peer receipt is known before the next message is sent, but again, this is slow because of the connection latency. TCP was not designed with this approach in mind.
If the commonly available IP protocols cannot directly provide what you need, you will have to resort to implementing your own.
What sort of 'packet' are you sending?
Rgds,
Martin
With TCP sockets, you just see the datastream where you can receive and send bytes. You have no way of knowing where a packet ends and another begins.
This is a feature (and a problem) of TCP. Most people just read data into a buffer until a linefeed (\n) is seen. Then process the data and wait for the next line. If transferring chunks of binary data, one can first inform the receiver of how many bytes of data are coming.
If packet boundaries are important, you could use UDP but then the packet order might change or some packets might be lost on the way without you knowing.
The newer SCTP protocol behaves much like TCP (lost packets are resend, packet ordering is retained) but with SCTP sockets you can send packets so that receiver gets exactly the same packet.
I'm implementing a remote application. The server will process & render data in real time as animation. (a series of images, to be precise) Each time, an image is rendered, it will be transferred to the receiving iPhone client via UDP.
I have studied some UDP and I am aware of the following:
UDP has max size of about 65k.
However, it seems that iPhone can only receive 41k UDP packet. iPhone seems to not be able to receive packet larger than that.
When sending multiple packets, many packets are being dropped. This is due to oversizing UDP processing.
Reducing packet size increase the amount of packets not being dropped, but this means more packets are required to be sent.
I never write real practical UDP applications before, so I need some guidance for efficient UDP communication. In this case, we are talking about transferring rendered images in real time from the server to be displayed on iPhone.
Compressing data seems mandatory, but in this question, I would like to focus on the UDP part. Normally, when we implement UDP applications, what can we do in terms of best practice for efficient UDP programming if we need to send a lot of data non-stop in real time?
Assuming that you have a very specific and good reason for using UDP and that you need all your data to arrive ( i.e. you can't tolerate any lost data ) then there are a few things you need to do ( this assumes a uni-cast application ):
Add a sequence number to the header for each packet
Ack each packet
Set up a retransmit timer which resends the packet if no ack recv'ed
Track latency RTT ( round trip time ) so you know how long to set your timers for
Potentially deal with out of order data arrival if that's important to your app
Increase receive buffer size on client socket.
Also, you could be sending so fast that you are dropping packets internally on the sending machine without them even getting out the NIC onto the wire. On certain systems calling select for write-ablity on the sending socket can help with this. Also, calling connect on the UDP socket can speed up performance leading to less dropped packets.
Basically, if you need guaranteed in-order delivery of your data than you are going to re-implement TCP on top of UDP. If the only reason you use UDP is latency, then you can probably use TCP and disable the Nagle Algorithm. If you want packetized data with reliable low latency delivery another possibility is SCTP, also with Nagle disabled. It can also provide out-of-order delivery to speed things up even more.
I would recommend Steven's "Unix Network Programming" which has a section on advanced UDP and when it's appropriate to use UDP instead of TCP. As a note, he recommends against using UDP for bulk data transfer, although the reality is that this is becoming much more common these days for streaming multimedia apps.
Small packets is probably better than large packets :-)