I have an android app that sends a request to the server via Volley.
The communication is fine but I see unnecessary packets(no.385 ~ no.387) as below. These packets are always sent 5 minutes after the last packet is sent. Please let me know why the client sends the packets to the server and how I can avoid sending these packets.
client : 10.x.x.x
server : 52.x.x.x
Packets 385-387 are relevant here:
image of my wireshark capture
UDP:
Datagrams – Packets are sent individually and are checked for
integrity only if they arrive. Packets have definite boundaries which
are honored upon receipt, meaning a read operation at the receiver
socket will yield an entire message as it was originally sent.
If a datagram can be split into multiple packets, it seems that there must be some timeout that the IP or UDP layer use to discard packets if not all packets for a datagram are received.
Is there a timeout? What is an example timeout?
I've got a little client program that listens on an IPv6 multicast group (e.g. ff12::blah:blah%en0) for multicast packets that are sent out by a server. It works well.
The server would also like to sometimes send a unicast packet to my client (since if the packet is only relevant to one client there is no point in bothering all the other members of the multicast group with it). So my server just does a sendto() to my client's IP address and the port that the client's IPv6 multicast socket is listening on.
If my client is running under MacOS/X, this works fine; the unicast packet is received by the same socket that receives the multicast packets. Under Windows, OTOH, the client never receives the unicast packet (even though it does receive the multicast packets without any problems).
My question is, is it expected that a multicast-listener IPv6 UDP socket should also be able to receive unicast packets on that same port (in which case perhaps I'm doing something wrong, or have Windows misconfigured)? Or is this something that "just happens to work" under MacOS/X but isn't guaranteed, so the fact that it doesn't work for me under Windows just means I had the wrong expectations?
It should work fine. As long as you bind to IN6ADDR_ANY, then join the multicast groups, you should be able to send and receive unicast packets with no problem.
It's important to bind to IN6ADDR_ANY (or INADDR_ANY for IPv4) when using multicast. If you bind to a specific interface, this breaks multicast on Linux systems.
I’m writing a simple client-server app which for the time being will be for my own personal use. I’m using Winsock for the net communication. I have not done any networking for the last 10 years, so I am quite rusty. I’d like to use as little external code as possible, so I have written a home-made server discovery mechanism, as follows.
The client broadcasts a message containing the ‘name’ of a client UDP socket bound to an arbitrary port, which I will call the client’s discovery socket. The server recv() the broadcast and then sendto() the client discovery socket the ‘name’ of its listening socket. The client then uses this info to connect to the server (on a different socket). This mechanism should allow the server to bind its listening socket to the first port it can within the dynamic port range (49152-65535) and to the clients to discover where the server is and on which port it is listening.
The server part works fine: the server receives the broadcast messages and successfully sends its response.
On the client side the firewall log shows that the server’s response arrives to the machine and that it is addressed to the correct port (to the client’s discovery socket).
But the message never makes it to the client app. I’ve tried doing a recv() in blocking and non-blocking mode, and there is never any data available. ioctlsocket() always shows no data is available, even though I know the packet got it to the machine.
The server succeeds on doing a recv() on broadcasted data. But the client fails on doing a recv() of the server’s response which is addressed to its discovery socket.
The question is very vague: what gotchas should I watch for in this scenario? Why would recv() fail to get a packet which has actually arrived to the machine? The sockets are UDP, so the fact that they are not connected is irrelevant. Or is it?
Many thanks in advance.
The client broadcasts a message containing the ‘name’ of a client UDP socket bound to an arbitrary port, which I will call the client’s discovery socket.
The message doesn't need to contain anything. Just broadcast an empty message from the 'discovery socket'. recvfrom() will tell the server where it came from, and it can just reply directly.
The server recv() the broadcast and then sendto() the client discovery socket the ‘name’ of its listening socket.
Fair enough, although actually the server could just broadcast its own TCP listening port every 5 seconds or whatever.
On the client side the firewall log shows that the server’s response arrives to the machine and that it is addressed to the correct port (to the client’s discovery socket). But the message never makes it to the client app
If it got to the host it must get to the application. You must have got the ports mixed up somehow. Simplify it as above and retry.
Well, it was one of those stupid situations: Windows Firewall was active, besides the other firewall, and silently dropping packets. Deactivating it solved the problem.
But I still don’t understand how it works, as it was allowing the server to receive packets sent through broadcasting. And when I got at my wits end and set the server to answer back through a broadcast, THOSE packets got dropped.
Two days of frustration. I hope someone profits from my experience.
This morning, there were big problems at work because an SNMP trap didn't "go through" because SNMP is run over UDP. I remember from the networking class in college that UDP isn't guaranteed delivery like TCP/IP. And Wikipedia says that SNMP can be run over TCP/IP, but UDP is more common.
I get that some of the advantages of UDP over TCP/IP are speed, broadcasting, and multicasting. But it seems to me that guaranteed delivery is more important for network monitoring than broadcasting ability. Particularly when there are serious high-security needs. One of my coworkers told me that UDP packets are the first to be dropped when traffic gets heavy. That is yet another reason to prefer TCP/IP over UDP for network monitoring (IMO).
So why does SNMP use UDP? I can't figure it out and can't find a good reason on Google either.
UDP is actually expected to work better than TCP in lossy networks (or congested networks). TCP is far better at transferring large quantities of data, but when the network fails it's more likely that UDP will get through. (in fact, I recently did a study testing this and it found that SNMP over UDP succeeded far better than SNMP over TCP in lossy networks when the UDP timeout was set properly). Generally, TCP starts behaving poorly at about 5% packet loss and becomes completely useless at 33% (ish) and UDP will still succeed (eventually).
So the right thing to do, as always, is pick the right tool for the right job. If you're doing routine monitoring of lots of data, you might consider TCP. But be prepared to fall back to UDP for fixing problems. Most stacks these days can actually use both TCP and UDP.
As for sending TRAPs, yes TRAPs are unreliable because they're not acknowledged. However, SNMP INFORMs are an acknowledged version of a SNMP TRAP. Thus if you want to know that the notification receiver got the message, please use INFORMs. Note that TCP does not solve this problem as it only provides layer 3 level notification that the message was received. There is no assurance that the notification receiver actually got it. SNMP INFORMs do application level acknowledgement and are much more trustworthy than assuming a TCP ack indicates they got it.
If systems sent SNMP traps via TCP they could block waiting for the packets to be ACKed if there was a problem getting the traffic to the receiver. If a lot of traps were generated, it could use up the available sockets on the system and the system would lock up. With UDP that is not an issue because it is stateless. A similar problem took out BitBucket in January although it was syslog protocol rather than SNMP--basically, they were inadvertently using syslog over TCP due to a configuration error, the syslog server went down, and all of the servers locked up waiting for the syslog server to ACK their packets. If SNMP traps were sent over TCP, a similar problem could occur.
http://blog.bitbucket.org/2012/01/12/follow-up-on-our-downtime-last-week/
Check out O'Reilly's writings on SNMP: https://library.oreilly.com/book/9780596008406/essential-snmp/18.xhtml
One advantage of using UDP for SNMP traps is that you can direct UDP to a broadcast address, and then field them with multiple management stations on that subnet.
The use of traps with SNMP is considered unreliable. You really should not be relying on traps.
SNMP was designed to be used as a request/response protocol. The protocol details are simple (hence the name, "simple network management protocol"). And UDP is a very simple transport. Try implementing TCP on your basic agent - it's considerably more complex than a simple agent coded using UDP.
SNMP get/getnext operations have a retry mechanism - if a response is not received within timeout then the same request is sent up to a maximum number of tries.
Usually, when you're doing SNMP, you're on a company network, you're not doing this over the long haul. UDP can be more efficient. Let's look at (a gross oversimplification of) the conversation via TCP, then via UDP...
TCP version:
client sends SYN to server
server sends SYN/ACK to client
client sends ACK to server - socket is now established
client sends DATA to server
server sends ACK to client
server sends RESPONSE to client
client sends ACK to server
client sends FIN to server
server sends FIN/ACK to client
client sends ACK to server - socket is torn down
UDP version:
client sends request to server
server sends response to client
generally, the UDP version succeeds since it's on the same subnet, or not far away (i.e. on the company network).
However, if there is a problem with either the initial request or the response, it's up to the app to decide. A. can we get by with a missed packet? if so, who cares, just move on. B. do we need to make sure the message is sent? simple, just redo the whole thing... client sends request to server, server sends response to client. The application can provide a number just in case the recipient of the message receives both messages, he knows it's really the same message being sent again.
This same technique is why DNS is done over UDP. It's much lighter weight and generally it works the first time because you are supposed to be near your DNS resolver.