Using MAVLink Router, I created three local host endpoints (bottom left window of the image) and I am able to see the live data through QGroundControl ports 14550 in the MAVLink Inspector, but I can't seem to grab the data in my listen.py script. I don't see anything on wireshark going to the specified ports, but I think it may be because it comes in through the sik telemetry radio in mavlink protocol.
The listen script
The bottom left is my onboard computer running mavlink-router
OP has probably solved this by now, but answering for others having this problem. In your listener script you passed a connection string "udp:localhost:14552". Your mavink router instance indicates that the UDP endpoints are clients. This means both your listener and your mavlink router udp endpoints are both listening for mavlink packets. Mavlink router will not send mavlink packets to those client endpoints until they receive mavlink heartbeats from the expected ip and port.
You should change your listeners pymavlink connection string to "udpout:localhost:14552 and have your script sending mavlink heartbeats either in a thread or just sending them every second in an infinite loop.
TL;DR Set your connection string to udpout so it is a "udp server" and send mavlink heartbeats. Mavlink router should start responding with messages as soon as those heartbeats come in.
Alternatively, you could set your mavlink router endpoints to be "udp server" endpoints by changing your mavlink-router conf file. Type "mavlink-routerd --help" to see how you can set up endpoints to "listen" or send mavlink messages without first receiving a heartbeat.
Hope that helps
Related
The goal is to make an http request from the client browser to my server. Simple stuff; however I'm hitting a wall with the networking portion. In order to expose my server to WAN I have used one of my public IPs and NAT to translate to the private ip of my server on inbound traffic and to my public IP on outbound traffic.
The issue is that I can't make a connection. Specifically I can't get the last part of the TCP handshake. Using a test setup with Wireshark on the client and server I can see that the client send the SYN -> the server receives the SYN -> the server sends a SYN/ACK -> the client receives a SYN/ACK -> the the client send an ACK -> the server DOES NOT receive the ACK. It waits for a moment then does a retransmission. Eventually resetting.
I have tried adding various firewall rules even though I don't think it could be the firewall because the first packets make a successful round trip.
I've turned windows firewall off(the server)
I've tried disabling TCP checksum offloading
I've looked for network anti virus settings on the server and on the sonic wall(the router)
I would expect the the tcp connection to complete. I can't for the life of me think of a reason why the ACK would consistently go missing.
That is another thing. The behavior is consistent.
pings also work just fine.
NOTE: The server is actually a VM and the physical server that manages it is in my network.
Any guidance on what to try and where to look would be very much appreciated. Thanks.
UPDATE: I can make connection using port 5000(It's another port I have opened on the firewall). Port 80 still doesn't work though.
In my case this was caused by COX not allowing inbound traffic to port 80. I'm not sure why the first portions of the tcp handshake were getting through. If anyone can explain that part leave a comment.
Setup:
The user has two applications - one sender one receiver - running on the same host/server. The user sets it up such that the sender sends messages to its own IP address not 127.0.0.1. Lets say its IP and port is x:y for simplicity. The user then sets up the receiver to receiver messages on x:y. Again this is on the same host/server.
Questions:
From my understanding this is not possible since the port will already be reserved. Therefore I cannot use the same port to try and send packets out to myself. Can I have a port used for a sender and receiver on the same node?
Is this resolved if I use SO_REUSEADDR or does this only resolve the IP conflict and not the port reuse?
If the program is not setup with IP_MULTICAST_LOOP the host will not multicast the message to itself, correct?
With IP_MULITCAST_LOOP set, if I only wanted to send the message to myself can I use 127.0.0.1 or must I use another address? Additionally, how do the ports get resolved?
If I am not seeing messages on the same node, would the first best guess be that IP_MULITCAST_LOOP is not set?
Let's take it step by step:
The sending port does not matter at all. So you can choose an arbitrary port for the sender, and use the specific port number for your service just for the receiver.
No, SO_REUSEADDR/PORT does not solve this problem. Even if you manage to achieve it: Do not start multiple listeners on the same port. This will cause strange effects. The main purpose of SO_REUSEADDR/PORT is to allow servers to create a TCP (not UDP) socket when the previous server process just died, without waiting for a timeout of the TCP state machine of the stale socket.
Corrects, assuming you mean multicast rather than broadcast,
Yes and no: If you only want to send messages to yourself you can send the packets to 127.0.0.1, and then you message will be a normal unicast packet and no longer a multicast packet, and IP_MULTICAST_LOOP does not matter at all. Multicast packets are normal UDP packets which have a destination address in the multicast address range (i.e. 224.0.0.0-239.255.255.255). The receiving socket cannot easily tell whether a packet was sent via unicast or multicast.
IP routing on the same host between interfaces is far from trivial. There are a lot of mechanisms and routing rules involved which are not shown in the normal routing table, which is just for outgoing traffic. It also depends on by which means you try to observe the messages. There is not a single point where you can see all messages going through a node (unfortunately). This is usually all attached to interfaces, and there also to an ingress and egress side, and the latter is usually not documented and not configurable. Monitoring local traffic can be tricky and may require virtual network interfaces. Really messy.
In summary: You are trying to send messages from one process to another process on the same host. Use unicast UDP for this and you are done. No multicast involved.
Can someone describe from a network point of view what RPC (SUN and/or DCE) is and why it deviates from standard TCP behavior?
The way that I understand it is a client reaches out to a server with a unique source port and then switches the source port after the TCP three way handshake finishes. I work with ASA firewalls so this behavior becomes very apparent when the inspection of DCE RPC is not enabled since the firewall will block it because it sees it as a threat. I have read a few MS TechNet articles and other website definitions to include watching about five Youtube videos which all seem to explain it from a programmers perspective but I have yet to fully understand this concept since I am not a programmer.
Note that there is nothing that deviates from standard TCP regarding the RPC protocols.
SunRPC or DCE RPC works on top of UDP(at least SunRPC can use UDP) or on top of TCP.
Typically in order for an RPC client to contact/call an RPCserver, it first contacts some sort of lookup server (called portmapper or rpcbind in the case of SunRPC), which replies with the location (IP address and port number) of where the actual server is running.
So from a networking perspective:
RPC Servers listens on a random port number, which may change each time that server program is (re)started.
At startup the RPC server connects to the portmapper, which runs on a well known port and register itself with which IP address and port number it's listening on.
Normally the portmapper service runs on the same machine as the RPC server programs.
When a client wants to connect to or call an RPC service it performs these actions:
Connects to the portmapper, on a well known/standard destination port and asks it where the particular service it wants to connect to is.
portmapper replies with the IP address and port number of the service the client asked for.
client tears down the connection to the portmapper
client establishes a new connection to the service using the IP address and port number that pormapper gave it.
client calls the RPC service over this new connection, which the client may use for multiple RPC calls.
These RPC calls are just application message exchanged on top of a TCP connection.
(In the case UDP is used instead of TCP, it works much the same, but there's no naturally no connection setup/teardown being performed over the network)
This presents a problem for firewalls, since the servers listens on randomly chosen port numbers, one cannot administratively allow access to a particular port number. Instead a firewall wanting to support this kind of setup would need to open up the portmapper port, catch the RPC messages going to that well known port of portmapper, inspect the message content exchanged with the portmapper to extract the IP address and port number from the RPC messages(the portmapper is itself implemented as an RPC server) in order to dynamically open a port between the RPC server and client.
Does the client remote port changes during an HTTP 1.1 connection exchange?
I am trying to figure out if I can programmaticaly uniquely identify a connection on the server using the request remote port and remote ip address.
This is not as much an HTTP question, as it's a TCP one. And no, the port doesn't change: the ephemeral port stays the same for the duration of the connection.
However, as soon as a new connection is made, the client can (and most probably will) use a different port. This totally depends on the implementation of the client OS and the Network Address Translation of intermediary routers.
Anyway, it is not something you can depend on to build something like a session, because the next request from the same client may very well arrive from a different port (let alone that HTTP does not have to run on top of TCP).
Just use a session-ID which you store in a cookie.
What I think of a port is: Whenever a message arrives to a machine, it is copied to a memory area which is mapped to the port specified and the concerned application or service is notified that a message has arrived for it.
If this is true, then what happens if two messages arrive for two different services listening on the same port ? ( either http or tcp )
And why can not two named pipe addresses use the same named pipe ?
TCP identifies "connections" via a tuple of { local ip, local port, remote ip, remote port }. Therefore, since each incoming connection has a different remote ip/port pair, your local machine can distinguish between them.
HTTP uses TCP for its transport. Thus, an HTTP port is a TCP port.
If you've ever had your machine get a new IP address while you had connections open, you'll note that they break the first time they send any data out since the remote host does not recognize the (new) address and sends a RST response.
A pipe has only its name to distinguish it so there is only one "connection" no matter how many writers it has.
Your description is one way to handle incoming messages.
In the case of two web sites listening on the same port, there is one web server listening on that port, which then looks at the http host header to find the correct web site to forward the request to.
The same is true for named pipes, the RPC listener listens on the TCP port, and then finds out that it is a named pipe message and then forwards the message to the right named pipe.