I was reading about RPC. The blog,
https://www.cse.iitk.ac.in/users/dheeraj/cs425/lec26.html, recommends to use UDP over TCP, in making remote procedure call, why is UDP preferred than TCP?
UDP is not generally preferred to TCP when doing remote procedure calls. In fact, most implementations of RPC technologies like CORBA, XML-RPC, SOAP, Java RMI, ... use TCP and not UDP as underlying transport. TCP is preferred here because contrary to UDP it already cares about reliability (dealing with packet loss, duplication, reordering) and can also easily and transparently handle arbitrary sized messages.
The blog you cite refers to classic Sun-RPC as used with NFS and which was primarily used in a local network - contrary to current RPC technologies which are often used in more complex network environments. In this kind of environment and at this time (long ago) UDP offered a smaller overhead and a faster recovery from network problems than TCP since there is no initial handshake and necessary retransmits, reordering ... are in full control of the RPC layer and can be tuned to the specific use case. So while preferring UDP for specific RPC in this environment made sense it cannot be said that UDP should be preferred for any kind of RPC.
Related
So I have a PC connected to a micro-controller via a serial cable and an Ethernet cable. Initially the PC sends a byte across the serial cable to the micro-controller. This results in the micro-controller sending back a UDP datagram via the Ethernet cable.
I want to know whether the code running on my PC should be a server or a client?
Per Wikiepdia Client/Server:
The server component provides a function or service to one or many
clients, which initiate requests for such services
And Master/Slave:
Master/slave is a model of asymmetric communication or control where
one device or process controls one or more other devices or processes
and serves as their communication hub
The above scenario looks like the Master/Slave. In the initial, 'idle' case, there is no "SERVER" that is waiting ("listening") for requests. Only when the PC activate the micro-controller they will start communication (via UDP).
You could use either term depending on what you were talking about. As other people have noted, client and server are terms used to describe how distinct parties are involved in a service. The terms can be useful in some situations (e.g. a web server and the browser as a client) but in other situations it's a less useful term (e.g. peer-to-peer protocols).
Presumably you're on stackoverflow because you're dealing with code.
In this case it's useful to be more precise and I'd suggest using terms to match whatever primitives are exposed by your language. Most will use/expose Posix sockets as their standard API, and hence you'd want to talk about/use connect or accept (potentially after binding first). Note that these calls work across TCP and UDP (except accept), but the semantics of sending and recving on the resulting connected sockets will obviously be different.
I'm building a LoRa network where the server and the end-device need to communicate using a protocol which normally transmits data via UDP. Due to the fact that these two protocols act totally different I need to find a way to combine those two.
One solution I found is to create my own socket API which provides send, receive, bind, ... functions. But here I'm actually struggling.
In which scope do I need to write my socket? Is it enough to just edit the functions and rely on the other given parameters such as the address families? Or must I define my own AF and if so where/how is this achieved.
I'm looking forward to your answers / ideas.
According to the LoRaWAN specification and my limited experience, LoRaWAN is not suitable for such situation. If you still wanna use UDP packets over LoRaWAN, here are some tips for your question.
In which scope do I need to write my socket?
You may use sendUnconfirm function since this function does not need ACK from gateway. And port in LoRaWAN could play the role of bind in UDP socket.
Is it enough to just edit the functions and rely on the other given
parameters such as the address families?
LoRaWAN server has its own features and structure. Usually, a LoRaWAN server is consist of packet_forwarder, LoRaWAN server and LoRaWAN application server. You may use these features to build you own application on LoRaWAN applicaiton server. It could save your a lot of time.
It is highly recommended to read LoRaWAN specification (Get it here) and TTN LoRaWAN wiki to help you get a better understanding in LoRaWAN.
Consider the prototypical multiplayer game server.
Clients connecting to the server are allowed to download maps and scripts. It is straightforward to create a TCP connection to accomplish this.
However, the server must continue to be responsive to the rest of the clients via UDP. If TCP download connections are allowed to saturate available bandwidth, UDP traffic will suffer severely from packet loss.
What might be the best way to deal with this issue? It definitely seems like a good idea to "throttle" the TCP upload connection somehow by keeping track of time, and send() on a regular time interval. This way, if UDP packet loss starts to occur more frequently the TCP connections may be throttled further. Will the OS tend to still bunch the data together rather than sending it off in a steady stream? How often would I want to be calling send()? I imagine doing it too often would cause the data to be buffered together first rendering the method ineffective, and doing it too infrequently would provide insufficient (and inefficient use of) bandwidth. Similar considerations exist with regard to how much data to send each time.
It sounds a lot like you're solving a problem the wrong way:
If you're worried about losing UDP packets, you should consider not using UDP.
If you're worried about sharing bandwidth between two functions, you should consider having separate pipes (bandwidth) for them.
Traffic shaping (which is what this sounds like) is typically addressed in the OS. You should look in that direction before making strange changes to your application.
If you haven't already gotten the application working and experienced this problem, you are probably prematurely optimizing.
To avoid saturating the bandwidth, you need to apply some sort of rate limiting. TCP actually already does this, but it might not be effective in some cases. For example, it has no idea weather you consider the TCP or UDP traffic to be the more important.
To implement any form of rate limiting involving UDP, you will first need to calculate UDP loss rate. UDP packets will need to have sequence numbers, and then the client has to count how many unique packets it actually got, and send this information back to the server. This gives you the packet loss rate. The server should monitor this, and if packet loss jumps after a file transfer is started, start lowering the transfer rate until the packet loss becomes acceptable. (You will probably need to do this for UDP anyway, since UDP has no congestion control.)
Note that while I mention "server" above, it could really be done either direction, or both. Depending on who needs to send what. Imagine a game with player created maps that transfer these maps with peer-to-peer connections.
While lowering the transfer rate can be as simple as calling your send function less frequently, attempting to control TCP this way will no doubt conflict with the existing rate control TCP has. As suggested in another answer, you might consider looking into more comprehensive ways to control TCP.
In this particular case, I doubt it would be an issue, unless you really need to send lots of UDP information while the clients are transferring files.
I wold expect most games to just show a loading screen or a lobby while this is happening. Neither should require much UDP traffic unless your game has it's own VOIP.
Here is an excellent article series that explains some of the possible uses of both TCP and UDP, specifically in the context of network games. TCP vs. UDP
In a later article from the series, he even explains a way to make UDP 'almost' as reliable as TCP (with code examples).
And as always... and measure your results. You have no way of knowing if your code is making the connections faster or slower unless you measure.
"# If you're worried about losing UDP packets, you should consider not using UDP."
Right on. UDP means no guarentee of packet delivery, especially over the internet. Check the TCP speed which is quite acceptable in modern day internet connections for most users playing games.
I believe that my question is:
Is there a simple user-mode TCP stack on PC operating systems that could be used to exchange data over a lossy serial link with a Linux-based device?
Here is more context:
I have a Linux-based device connected via a serial link to a PC. The serial link is lossy so data being sent between the two devices sometimes needs to be retransmitted. Currently the system uses a custom protocol that includes framing, addressing (for routing to different processes within the Linux device), and a not-so-robust retransmission algorithm.
On the Linux device side, it would be convenient to replace the custom protocol, implement SLIP over the serial link and use TCP for all communications. The problem is that on the PC-side, we're not sure how to use the host's TCP stack without pulling in general IP routing that we don't need. If there were a user-mode TCP stack available, it seems like I could integrate that in the PC app. The only TCP stacks that I've found so far are for microcontrollers. They could be ported, but it would be nice if there were something more ready-to-go. Or is there some special way to use the OS's built in TCP stack without needing administrative privileges or risking IP address conflicts with the real Ethernet interfaces.
Lastly, just to keep the solution focused on TCP, yes, there are other solutions to this problem such as using HDLC or just fixing our custom protocol. However, we wanted to explore the TCP route further in case it was an option.
It appears that the comments have already answered your question, but perhaps to clarify; No you can not use TCP without using IP. TCP is built on top of IP, and it isn't going to work any other way.
PPP is a good way of establishing an IP connection over a serial link, but if you do not have administrative access on both sides of the computer this could be difficult. 172.16.x, 10.x, and 192.168.x are defined as being open for local networks, so you should be able to find a set of IP addresses that does not interfere with the network operation of the local computer.
From the point of view of no configuration, no dependencies, comping up with your own framing / re-transmit protocol should not be too hard, and is probably your best choice if you don't need inter-operability. That being said kermit, {z,y,z}modem would provide both better performance and a standard to code against.
Lastly, you may be able to use something like socat to do protocol translation. I.e. connect a serial stream to a TCP port. That wouldn't address data reliability / re-transmission, but it may be the interface you are looking to program against.
I am unable to understand or grasp rather; what it means to program at a lower layer in socket programming. I am used to working with tcp/udp/file system sockets. These are all wrapped around their own protocol specifications ... which as i understand would make it working at the application layer in the stack.
In the project i am on , i have seen some files which are "named" LinkLayer, TransportLayer... but i don't see any more calls other than standard socket calls....send /recv/ seletct...
Does the fact you are setting a socket options mean you are programming at a lower level ? Is it just restricted to it? Or are there other API's which grant you access at the representation in kernel ?
Typically this refers to using SOCK_RAW sockets, which requires you to assemble your own packet headers, calculate checksums, etc. You still use send/recv/etc. but now you are responsible for making sure every bit is in the right place.
You can use SOCK_RAW sockets to implement protocols other than TCP or UDP, or to do things with the Internet protocols that higher-level interfaces don't accommodate (like tweaking the TTL of your packets to implement something like traceroute).
This usually means working on a lower OSI-Layer, for example, not directly sending TCP-streams or UDP-packets, but crafting own IP or even Ethernet packets or other low-layer protocols which would - in normal case - be handled by the operating system.
This can be done done via specific socket options which enable you to receive or send data on any layer, even layer 2 (Data Link).