Develop Reference

Need of checksum in serial communication

I am currently working on a project involving a Serial COM from PC ( USB TO SERIAL application coded in C# ) to an embedded platform (STM32F4).
I saw that in some cases it's mandatory to have a checksum in a communication frame.
The Communication configuration:
Baud-rate = 115200,
No Parity bit,
One StopBit,
No Handshake,
Frame Length : 16 bytes
Is it worth adding a checksum in my application? What are the reasons why i should (not) have this checksum?
Thank you for your answer.

Yes you must have a checksum. The only acceptable non-hobbyist solution is a proper checksum based on CRC. Most common industry standard is "CRC-16-CCITT" (polynomal 0x1021). This will catch any single-bit error, most double-bit errors and some burst errors.
Even though you'll only use RS-232 in an office environment(?), any EMI caused by crappy consumer electronics could cause glitches and incorrect data. There is a lot of such crappy electronics around: for example it is not all that uncommon for the electronics in your PC to have poor EMC performance. In particular, there are countless of USB-to-serial adapters with downright awful quality.
The UART hardware in itself has no error detection worth mentioning: it is ancient 1960s technology. On the hardware level, it only checks data integrity based on start and stop bits, and will miss out any errors in between. (Parity checking is equally poor).
Alternatively, you could perhaps get an USB to RS-485 adapter instead and use RS-485, which is far more rugged as it has differential signals. But that requires that you have RS-485 transceivers on the target side too.

It is customary to have a checksum in order to verify the correctness of the data although serial communication is relatively reliable. You will definitely need a sync made up of at least two bytes that will always be assigned a specific value which you don't expect to appear in your data. The sync is used in the receiving side to find the start of each message sent because it is a stream communication and not a packet based communication.

Reliability over a serial connection

I have two computers communicating over a serial modem.
I would like to have a reliability protocol on that line.
I have been looking into PPP, SLIP and RATP. Not all of them are the best fit, and I do not want to write all that code, especially if there is a good code base for that online.
Is there a library or code project in C# that can be used for that purpose?
If not what protocol should you recommend to implement?
The connection speed is 9600, but the amount of data sent is not very big, and speed is not a big issue. Simplicity and ease of development is!

I always just add a CRC to each message, but my higher level protocols are self-synchronizing and loss tolerant by virtue of unsolicited state reports -- if a command is lost and the state doesn't change, that becomes apparent on the next state report. Depending on whether your requirement is to detect or correct errors, and whether you can tolerate extra delays for retransmissions, you might need to look into a forward error correcting code.
Concepts of interest include message sequence numbers, acknowledgement, go-back-N vs selective retransmit, and minimum distance between codes (Hamming distance).
I definitely suggest you look at the design of TCP. The basics are really pretty minimal for guaranteed in-order delivery.

Record and replay UDP packets with timing

I need to implement a way of recording an UDP stream with the purpose of later on replaying that stream as requested. This record and replay must use the same timing (up to a sensible resolution, hardly perceivable by a human user, let's say 10ms): if the original stream had data for seconds 00:00 to 00:35, then went mute till 00:55, and then sent more data from 00:55 to 01:34, the same distribution must be observed when my application replays that stream.
If it were just a matter of saving the udp stream to disk and then replaying it, without this timing, it would be trivial by either using Socket, NetworkStream or UdpClient. The issue is I cannot get around to devise a way to modify the standard receive algorithm to include that timing in a way that it's somehow easily replayed later by the send algorithm. As a bonus, a way to start the replay at any time mark (from 00:15 on, for example) should also be supported.
1) Is there any way to easily implement this functionality on C#? We do not have any severe non-functional requirement for this, we just need for it to simply work. Any hint about a way to implement it would be really appreciated.
2) If this is indeed not a simple matter and anyone suggests the use of any third-party component for this, the requirements would be for it to have an API for C# (or a way to operate the component from code), and hopefully be opensource or freely used software.
Thank you very much.

This is not exactly a native feature of C#. But I will attempt to do your homework.
The way I would attack this is still to store the packets to disk. Design a custom file structure in binary format, Holding data about the time the Datagram was recieved followed by the UDP Datagram itself. You can then have a program read this file back, Find out at what time the UDPs payload was delivered. This will give you what you need to replay the packets and store it for later replay with all native C# Code, no third party modules. This will of course require you to know writing to/from a file and being able to parse the relevant data into the typed objects in C# (Timespan, maybe byte[] for the Datagram, etc). All of these decisions would be up to you, the program writer.
The long and the short of it is i'm 99% certain there's no native functionality for this kind of requirement. This is exactly Why programming languages exist. So we as programmers can produce amazing solutions for our clients/customers/Teachers ;)

UDP File Transfer - Yes, UDP

I have a requirement to create a UDP file transfer system. I know TCP is guaranteed and much more reliable, but I need to transfer huge files between locations and I think the speed advantage in this project outweighs the benefits using TCP. I’m just starting this project, but would like some guidance if anyone has done this before. I will be writing both sides (client and server) so I don’t need to worry about feature limitations in other products.
In a nutshell I need to:
Take large files and send them in chunks
Be able to throttle bandwidth from the client
Create some kind of packet numbering system for errors,
retransmitions and assembling files by chunk on server (yes, all the
stuff we get from TCP for free :-)
Configurable datagram size – I think some firewalls complain if they
get too big?
Anything else I may be missing
I’m starting this journey using UdpClient and would like to write this app in C#. Any words of wisdom (other than to use TCP)?
It’s been done with huge success. We used to use RocketStream.com, but they sold their product to another company for internal use only. We typically get speeds that are 30X faster than FTP or raw TCP byte transfers.

in regards to
Configurable datagram size – I think some firewalls complain if they get too big?
one datagram could be up to 65,536 bytes. cosidering all the ip header information you'll end up with 65,507 bytes for payload. but you have to consider how all devices are configured along you network path. typically most devices have set an MTU-size of 1500 bytes so this will be typically your limit "on the internet". if you set up a dedicated network between your locations you can increase your MTU an all devices.
further in regards to
Create some kind of packet numbering system for errors, retransmitions and assembling files by chunk on server (yes, all the stuff we get from TCP for free :-)
i think the best thing in your case would be to implement a application level protocol. like
32 byte sequence number
8 byte crc32 checksum (correct me on the bytesize)
any bytes left can be used for data
hope this gives you some bit of a direction
::edit::
from experience i can tell you UDP is about 10-15% faster than TCP on dedicated and UDP-tuned networks.

I'm not convinced the speed gain will be tremendous, but an interesting experiment. Such a protocol will look and behave more like one of the traditional modem based protocols, and probably ZModem is one of the better examples to get some inspiration from (implements an ack window, adaptive block size, etc).
There are already some people who tried this, check out this site.

That would be cool if you succeed.
Don't go in it without WireShark. You'll need it.
For the algorithm, I guess that you have pretty much the idea of how to start. Maybe some pointers:
start with MTU that will be common to both endpoints, and use packets of only that size, so you'll have control over packet fragmentation (when you come down from TCP, I hope that this is for the more control over low level stuff).
you'll probably want to look into STUN or TURN for punching the holes into NATs.
look into ZModem - that also has a nostalgic value :)
since you want to squeeze maximum from you link, try to put as much as you can in the 'control packets' so you don't waste a single byte.
I wouldn't use any CRC on packet level, because I guess that networks underneath are handling that stuff.

I just had an idea...
break up a file in 16k chunks (length is arbitrary)
create HASH of each chunk
transmit all hashes of the chunks, using any protocol
at receiving end, prepare by hashing everything you have on your hard drive, network, I mean everything, in 16k chunks
compare received hashes to your local hashes and reconstruct the data you have
download the rest using any protocol
I know that I'm 6 months behind the schedule, but I just couldn't resist.

Others have said more interesting things, but I would like to point out that you need to make sure you use a good compression algorithm. That will make a world of difference.
Also I would recommend validating your assumptions as to the speed improvement possibility, make a trivial system of sending data (not worrying about loss, corruption, or other problems) and see what bandwidth you get. This will at least give you a realistic upper bound for what can be done.
Finally consider why you are taking on this task? Will the speed gains be worth it after the amount of time spent developing it?

Tcp Reliability versus Udp Burdens for serious, high-performance server

Speed, optimization, and scalability are the typical comparisons between the Udp and Tcp protocols. Tcp touts reliability with the disadvantage of a little extra overhead, but speed is good to excellent. Once a Tcp socket is instanced, keeping the socket open requires some overhead. But compared to the oft described burdens of Udp, which protocol actually has more overhead?. I've also heard that there are scalability issues with Tcp...yet the Internet (Web pages/servers) runs on Tcp - so what is it about Tcp that inhibits scalability?
Okay...so Udp doesn't require that overhead of keeping a connection open. But, it requires that you write extra methods to ensure all of the packet gets there, hopefully in the order that you want it received. If a packet isn't received in full, then you have to tell the client or server to resend. And you also have to keep some sort of message collection for partial packets, rebuild the partial messages, and check for a complete message before the message can finally be processed. Not to mention if the second part of a message never makes it, you have to either say resend the entire thing, or resend the part we are missing, or whatever.
Basically, my questions are:
Why would I choose Udp over Tcp for a serious, high-performance server with the added "overhead" of message
checking and manual ACK versus the "overhead" of a continuous stream?
If Tcp is good enough for the likes of World of Warcraft, why isn't Tcp more widely accepted as the protocol to use for a game server?
Note: I am not opposed to implementing Udp options for a server. We are using C# on .Net 3.5 framework. So I would also be interested in the best practices for dealing with Udp burdens. I am also using the asynchronous methods at the socket level rather than using TcpListener, TcpClient, etc. etc.

Well, I would recommend reading up some more. There are plenty places to look at the pro's and con's of TCP vs. UDP and vice versa, here are a few:
What Are The Advantages Of Using TCP Over UDP?
When should I use UDP instead of TCP?
TCP and UDP
What are the advantages of UDP over TCP?
However, this link may interest you the most, as it is directly about networked game programming:
Gaffer on Games - UDP vs. TCP
If I were to quote something small:
The decision seems pretty clear then,
TCP does everything we want and its
super easy to use, while UDP is a huge
pain in the ass and we have to code
everything ourselves from scratch. So
obviously we just use TCP right?
Wrong.
Using TCP is the worst possible
mistake you can make when developing a
networked game! To understand why, you
need to see what TCP is actually doing
above IP to make everything look so
simple!
I still recommend doing your own research on the matter though, and make sure which of the protocols suits your needs at the end of the day. This being said, it does seem to be the case that majority of games use UDP for their data. Anything that updates the entire state continuously does not need the overhead of guaranteed packet delivery.

First, I'll just paraphrase Stevens from Unix Network Programming Section 22.4 "When to Use UDP instead of TCP":
He basically says the following:
UDP is the only option for broadcast / multicast - so you have to use it there.
UDP can be used for simple request / reply apps. But you have to add your own error detection meaning at least acks, timesouts and retransmission.
UDP should not be used for bulk data transfer ( file transfers ) since you would have to build in all the functionality arleady in TCP to make it work right.
UDP should be used for real time data where speed of delivery is most important and some data loss is not an issue such as real time sensor data, live multimedia streams, real time stock quotes, etc.
The answer to your first question is very dependent on your definition of "high-performance". If you're primary concern is low latency, i.e. the individual data packets / requests arriving as quickly as possible than UDP would be the way to go. There are two primary reasons for this. Assuming packets / requests are fairly independent of each other than using TCP would introduce a problem known as head-of-line blocking.
Let's say you send two independent packets / requests. First A then B. Since TCP is stream based, if A get's lost in the network and needs to be retransmitted then even if B has already successfully arrived it can't be delivered to the application by the stack until A arrives, introducing unnecessary latency. Not only that, but until A arrives, B can't be acknowledged by the stack which might cause B to also be retransmitted causing needless network congestion.
One way around this problem is to use separate connections for each request, however this also introduces latency and hogs system resources. UDP bypasses all these problems.
Another issue in high performance ( low latency ) servers is the Nagle Algorithm which can add significant latency in TCP communications.
The answer to your second question is that WoW probably sends streams of data, not independent request / reply pairs. Also, some of the latency of TCP can be removed by disabling the Nagle algorithm. If they do use some request / reply communications they may have simply made a design decision that reliability is more important to them than latency.

Define "serious high performance" - how many concurrent connections are you talking about and how much data is flowing?
Take a look at the answers to this question What do you use when you need reliable UDP? which list some of the reliable protocols that have already been built on UDP. You might find one that works for your situation, or you may at least find some useful ideas.
The key to using UDP effectively here is to have some level of reliability and some level of unreliability and you get more of an advantage the more each datagram is able to be handled independently of others. The advantage over TCP is that you get to act on each datagram and decide if you can use it as it arrives. This is why it works for action games.
So, IMHO, if you need 100% reliability AND in order delivery then go with TCP; don't try and reimplement TCP in UDP.

It's Reliability vs Performance.
FPS games don't require -all- the packets to reach the destination, to reach it in order, to be exceptionally big, or to assure big throughput. They only require the packets to reach the server AS SOON AS POSSIBLE. This is the ultimate priority and overhead of TCP is simply an unnecessary burden.
WoW, in its "not quite realtime" communication and often tons of data to transmit (in crowded areas), may have to deal with packets exceeding MTU (requiring fragmentation) and requires reliability (fewer bigger packets = packet lost hurts more). So its choice of TCP is logical. Same would go for most turn-based strategy games and the like. In games where the player with ping of 30ms beats the player with ping 50ms UDP is the king.

I think the biggest part of TCP/IP that inhibits scalability is that it maintains a buffer on all incoming / outgoing connections up to basically the size of the window. So if I have a high latency but high throughput client i'm talking to, I have to keep all sent packets in buffer until I receive an ack. So for a few connections this is fine, but for handling 100K connections, it can start to be problematic overhead. On the receiving end, if a packet is dropped, again it will buffer all new packets received until the one required is retransmitted.
If you're going to implement retransmission, you need to do the same thing, and hence will have the same overhead. However, UDP does give you an advantage, if you know the end-to-end link speeds, or if certain message can be delivered out of order, or certain messages don't need retransmission. Keeping the gaming scenario:
packet 1 = move to 1,1
packet 2 = shoot
packet 3 = move to 2,2
Most game designers, if packet 1 is lost, but packet 3 is received, packet 1 is no longer important because it contains out of date information anyways. However, you could opt to say packet 2 is important, so if it's not acked, send a retransmission.
If you need high throughput, and connect two servers directly with 1000Mbps ethernet, TCP/IP will take awhile to scale and have additional overhead, and will likely never achieve a true gigabit connection due to the congestion avoidance mechanisms. However, you know it's 1 Gbps, so you can set up you're UDP to transmit at up to a 1 Gbps (minus overhead) yourself.
To answer you're questions more directly:
If you are going to ack every packet anyways, there isn't a massive benefit to having UDP, other than you can process some messages while waiting for retransmission (unless you want in-order delivery as well).
Udp isn't considered for game servers as much, mainly out of the scenario above, and real time combat systems such as First person shooters, where a message can be dropped, and the new message to come will invalidate the dropped message anyways. World of warcraft can get away with using TCP, since they don't have to be as precise with timing, and likely have some good logic that makes it more difficult for you to tell the difference anyways. The combat system simply doesn't require the speed.
I'd also contend that some of the justification is holdover from years ago, when everyone had less-reliable, and slower Internet connections. TCP is also more lenient for sharing the network, so if there's a lot going on, it will slow down so everyone gets a share of the connection (congestion avoidance).
TCP/IP is a protocol designed by people far smarter than I over years of research. Tuning in the last several years has allowed it to perform better with the faster and faster average network speeds we are seeing, and doesn't require a great understanding to use.
However, replacing this with UDP, does require a significant understanding of networking. I've seen badly written UDP programs saturate 1Gbps links and kill all traffic on the link, because they implemented a rather naive retransmission algorithm.
Here's a list of things TCP/IP can now do that you'd loose by going UDP:
- In order arrival to you're program
- retransmission (Now with Fast retransmit, selective acknowledgement, and other features)
- Maximum segment size
- Path MTU Discovery
- Black Hole Detection (extension of Path MTU)
- Congestion avoidance
Because of this, I'd highly recommend sticking with TCP/IP if it suits you're needs.
Also not to nit pick, but you're comment about the Internet running on TCP/IP is wrong, there are in fact dozens of Internet routeable protocols check them out here. I think you were referring to web pages and web servers are all running on top of TCP/IP. Which again for the web is great where us humans won't notice a delay as long as the page shows up correctly. Even for TCP/IP, their is some challenge that TCP/IP isn't aggressive enough for the web: Google thinks tcp/ip should be more aggressive by default