I have recently developed a GPRS communication software using Arduino (embedded application) and GSM modem to communicate to/fro from web server. However I found that there is enough delay and request getting dropped (response timeout) while receiving a response from server at client side.
The techniques I have tried are - TCP / UDP / HTTPS / HTTP.
In my case our requirement is for a Reliable, Fast, Guarantee Communication between client and server.
Please let me know which communication stack would establish the same or rather be best to be used?
Thanks in advance
GPRS gives you direct IP access to the Internet. If you're losing packets or suffering large delays when sending packets to your server then this sounds like a problem with the mobile ISP.
As Ken mentioned GPRS will provide you IP connectivity to the internet (or some private network if applicable).
On top of IP you can choose to use a number of higher layer protocols, the two most common of which are probably UDP and TCP.
UDP is 'connectionless' and provides very little in the way of error detection/correction etc.
TCP is connection orientated (which means that some signalling happens back and forth to establish a virtual 'connection' first). It also includes mechanisms to provide error detection, error correction and correct packet delivery order. TCP also includes flow control, to avoid the sender overloading the receiver, and congestion control to avoid network overload.
There is a perception that UDP is faster than TCP, but I think it depends on the situation - take a look at this discussion for some further discussion on speed, reliability etc between UDP and TCP (go down through all the high scored answers):
UDP vs TCP, how much faster is it?
For your requirements, I would think a solution based on TCP/IP is probably what you want.
Whether you want to use HTTP or some other protocol on top of that is going to be dependent on your solution, and to some extent on personal preference.
Related
I would like to understand networking services with a large user base a bit better so that I know how to approach a project I am busy with.
The following statements that I make may be incorrect but they still lead to the question that I want to ask...
Please consider Skype and TeamViewer clients. It seems that both keep persistent network connections open to their respective servers. They use these persistent connections to initiate additional connections. Some of these connections are created by means of Hole Punching if the clients are behind NATs. They are then used for direct Peer-to-Peer communications.
Now according to http://expandedramblings.com/index.php/skype-statistics/ there are 300 million users using Skype and 4.9 million daily active users. I would assume that most of that 4.9 million users will most probably have their client apps running most of the day. That is a lot of connections to the Skype servers that are open at any given time.
So to my question; Is this feasible or at least acceptable? I mean, wouldn't it be better to not have a network connection open while idle and aspecially when there are so many connections open to the servers at once? The only reason I can think is that it would be the only way to properly do Hole Punching. Techically, how is this achieved on the server side?
Is this feasible or at least acceptable?
Feasible it certainly is, you mention already two popular apps that do it, so it is very doable in practice.
As for acceptable, to start no internet authority (e.g. IETF) has ever said it is unacceptable to have long-lived connections even with low traffic.
Furthermore, the only components for which this matters are network elements that keep connection/flow state. These are for sure the endpoints and so-called middleboxes like NAT and firewalls. For the client this is only one connection, the server is usually fine tuned by the application developers (who made this choice) themselves, so for these it is acceptable. For middleboxes it's simple: they have no choice, they're designed to just work with all kind of flows, including long-lived persistent connections.
I mean, wouldn't it be better to not have a network connection open while idle and aspecially when there are so many connections open to the servers at once?
Not at all. First of all, that could be 'much' slower as you'd need to set up a full connection before each control-plane call. This is especially noticeable if your RTT is big or if the servers do some complicated connection proxying/redirection for load-balancing/localization purposes.
Next to that this would historically make incoming calls difficult for a huge amount of users. Many ISP's block/blocked unknown incoming connections from the internet by means of a firewall. Similar, if you are behind a NAT device that does not support UPnP or PCP you can't open a port to listen on for your public IP address. So you need it even aside from hole-punching.
The only reason I can think is that it would be the only way to
properly do Hole Punching. Techically, how is this achieved on the
server side?
Technically you can't do proper hole-punching as soon as the NAT devices maintain a full <src-ip,src-port,dest-ip,dest-port,protocol> (classical 5-tuple) flow match. Then the best you can do with 'hole punching' is set up a proxy between peers.
What hole-punching relies on is that the NAT flow lookup is only looking at <src-ip,src-port,protocol> upstream and <dest-ip,dest-port,protocol> downstream to do the translation. In that case both clients just set up a connection to the server, their ip and port gets translated and the server passes this to the other client. The other client can now start sending packets to that translated <ip,port> combination which should work because NAT ignores the server's ip/port. But even if the particular NAT would work like this, some security device (e.g. stateful firewall) might detect session hi-jacking and drop this anyway.
Nowadays you rather use UPnP to open up a port to listen on your public IP which is much easier if supported.
I spent the last two days reading each StackOverflow questions and answers (and googling of course) about Indy TCP and UDP protocol in order to decide which one should I use in my communication method between my User Application and my Windows Service.
From what I saw so far, UDP is the easiest and the only one I managed to work to receive broadcast messages from TidUDPClient (I did not testes the response back yet). And I also noticed that TCP is a bit more complicated with it's thread loop.
But since everywhere I am told UDP is not reliable, UDP is not reliable... I begin to wonder if it's not better to use TCP anyway.
My User Application will be running on many machines, and the Service will be running in one of them, sharing one IP with a Client, or in a dedicated machine, depending on my client's funds. So, should I really be worried about UDP data loss possibilities?
I need broadcast capabilities so my server advises all clients at once about Application updates, and of course, if my the Client Application does not know in which IP the Service/Server is, it will send a broadcast call to be told where the server is. Is that applicable to TCP?
The messages I am sending are requests for users access confirmation, users privileges, and application executable file updates, since the main application can't update itself.
Those messages are encrypted like below, and they might bet bigger sometimes.
e86c6234bf117b97d6d4a0c5c317bbc75a3282dfd34b95446fc6e26d46239327f2f1db352b2f796e95dccd9f99403adf5eda7ba8
I decided to use them both!
Simple use case:
In order to communicate with TCP prococol you have to establish a connection which you can have only if you know IP and Port on both ends.
If you do not have that information when you load your Application, then you use the UDP to Broadcast your IP address and your intention to find the/a Server. You may try about 5 times before you raise the user an error telling that you did not find the Server or that the Server is down.
Sending that message in UDP will (one time or other) reach the UDP ear of the Server, which will now know the IP from the lonely Client's IP and will now begin a proper connection via TCP to be read talk about the critical messages of the Application.
What do you think of that approach?
I'm want to design a ruby / rails solution to send out to several listening sockets on a local lan at the exact same time. I want the receiving servers to receive the message at exact same time / or millisecond second level.
What is the best strategy that I can use that will effectively allow the receiving socket to receive it at the exact same time. Naturally my requirements are extremely time sensitive.
I'm basing some of my research / design on the two following articles:
http://onestepback.org/index.cgi/Tech/Ruby/MulticastingInRuby.red
http://www.tutorialspoint.com/ruby/ruby_socket_programming.htm
Now currently I'm working on a TCP solution and not UDP because of it's guaranteed delivery. Also, was going to stand up ready connected connections to all outbound ports. Then iterate over each connection and send the minimal packet of data.
Recently, I'm looking at multicasting now and possibly reverting back to a UDP approach with a return a passive response, but ensure the message was sent back either via UDP / TCP.
Note - The new guy syndrome here with sockets.
Is it better to just use UDP and send a broad packet spam to the entire subnet without guaranteed immediate delivery?
Is this really a time sensitive component? If it's truly down to the microsecond level then you may want to ensure its implemented close to native functions on the hardware. That being said a TCP ACK should be faster than a UDP send and software response.
1)
Now I am writing IM chat System i face some problem how to send vector that has information from the server to Client
2)
is any way to communicate between tow client ??
I Use CBC2010 - Indy10
Basically communicating over TCP is about sending bytes from client to server, and receiving bytes on the client from the server.
You either can give meaning to those bytes, or have something wrap that for you.
There are many possibilities and protocols to choose from.
On the foundation, you have either UDP (which is unreliable, but incurs almost no overhead, but very well suited for broadcasts) and TCP (which is more reliable, therefore has more overhead, but is easier to use).
A transport protocol that is often used on top of TCP is HTTP, especially since it is easy to get it through proxy servers.
On top of that you can do XML+SOAP or JSON+REST, which make translating from/to your underlying objects a lot easier.
All in all there are a truckload of options to choose from.
A simple start is the Delphi chat example at delphi.about.com. That definitely should get you going.
I am a newbie in network related aspects. I have few basic questions related to tcp/ip protocol and network
If a network switch (in a LAN network) between two PC's running Client and server (that are communicating through async. sockets) is powered down. Can the client and server will be notified that the socket connection is no longer active. Client and server are running on Win XP OS and are coded using C#.
Does network topology play a role in case of half open connection between socket client and socket server. For e.g. Will a disconnect status of either one or both be notified to other end and does it depend on network topology.
Thanks in advance.
A network element such as a router/hub/switch does not activly cause anything anything to happen on the TCP layer if it goes down. The operating system might notice that the physical layer is down and error out all sockets bound on that network card if it's a network element directly connected to the PCs that breaks - this will vary among operating systems/network cards and other things. Other than that, in order to detect that the connection has been severed, you'll have to send something and rely on the TCP timeout mechanisms to error out. This can be done implicittly by enabling TCP Keepalives on the connection.
A disconnect on one side will only be noticed if those messages reach the other side, if the network topology changes or sometinhg breaks in the middle of the connection in such a way that messages no longer reach the other end, a disconnect won't be noticed. (NAT gateways are a big source of problems such as this, they might time out a TCP connection they're tracking and you'll never know the connection is no longer valid unless you try to write something (or enable TCP keepalives) to the connection). Note that most networking APIs require that you Read from the connection to discovver that a the other end has closed the connection - assuming those "close" messages actually reach your side.