I want to use netsed to alter the incoming SIP traffic (UDP port 5060) on a PBX server which is running on a linux system (debian 10 stretch).
In the first step I simply tried to set up the phone to connect to UDP/5061 and started netsed with that command:
./netsed udp 5061 127.0.0.1 5060 s/profile-level-id=4280D/profile-level-id=42800D
The traffic was intercepted, changed and forwarded to port 5060 of the PBX software.
Unfortunately, the SIP protocol "notices" that the port used by the phone does not match the PBX (the pbx is using 5060 and the phone is using 5061 because of the port forwarding). Now the question is how to do this with netsed.
So the question is how to create a transparent proxy on the same system as the pbx.
Or is there a way to listen with netsed already on port 5060, but then pass it on to the pbx software on port 5060?
The deeper background is the Mobotix door phones for outgoing calls that do not correctly encode the profile-level-id field (it should be six base16 characters long but is five characters long). My idea was to simply change the profile-level-id as soon as the SIP-Invite enters the PBX.
Same question at StackExchange.
If I do not misinterpret your idea ; and I thing a quite interesting one ; you are trying to set up an transparent application IP proxy using netsed. From that experiment, the straightforward solution does not fit because SIP protocol "notices" that 'netsed' is altering the data-path.
Some points to be aware of :
By itself, SIP does not work via NAPT as the transferred data
contains IP addresses and port numbers.
Moreover taking aside TLS, SIP is transport protocol agnostic which
mean that a SIP session is authorized to use both UDP and TCP at the
same time.
I suggest you to have a look to SIP proxy, SBC B2BUA and STUN in order to build a correct understanding of all these issues, seem to be a hard path but valuable.
Or you can take inspiration of SIP proxy and implement minimun SIP proxying functionality into netsed. It is interesting but not so easy.
But if you abandon your initial idea of 'basic' proxy then you could use a SIP toolbox like [kamalio][1]. It is a known way of doing SIP related plumbing but obviously very far your initial goal.
Hope this help.
Related
I'm working on a project where some clients (embedded linux systems) needs to connect to a main server using so far at least two protocols: HTTPS and SSH. One of the requirement is that only one flow is allowed from every client to the server, so I've to found a way to make the two services works on the same port.
One solution that I was thinking about is to use the iptables markers: on the client side mark the packets for SSH with 0x1, the packets for HTTPS with 0x2 and then on the server side, based on the marker, redirect the packets to the right service listening on the local interface. Is it an acceptable solution? Are the markers kept by the network routers or is only something working locally on the same machine for iptables?
And anyway, if you can advice about a different solution, of course it's welcome!
More for other users finding this question in the future:
https://github.com/yrutschle/sslh has what you might need. I haven't used it (yet) but planning on it.
From the Github site:
sslh -- A ssl/ssh multiplexer
sslh accepts connections on specified ports, and forwards them further based on tests performed on the first data packet sent by the remote client.
Probes for HTTP, SSL, SSH, OpenVPN, tinc, XMPP are implemented, and any other protocol that can be tested using a regular expression, can be recognised. A typical use case is to allow serving several services on port 443 (e.g. to connect to SSH from inside a corporate firewall, which almost never block port 443) while still serving HTTPS on that port.
Hence sslh acts as a protocol demultiplexer, or a switchboard. Its name comes from its original function to serve SSH and HTTPS on the same port.
I'm writing a piece of P2P software, which requires a direct connection to the Internet. It is decentralized, so there is no always-on server that it can contact with a request for the server to attempt to connect back to it in order to observe if the connection attempt arrives.
Is there a way to test the connection for firewall status?
I'm thinking in my dream land where wishes were horses, there would be some sort of 3rd-party, public, already existent servers to whom I could send some sort of simple command, and they would send a special ping back. Then I could simply listen to see if that arrives and know whether I'm behind a firewall.
Even if such a thing does not exist, are there any alternative routes available?
Nantucket - does your service listen on UDP or TCP?
For UDP - what you are sort of describing is something the STUN protocol was designed for. It matches your definition of "some sort of simple command, and they would send a special ping back"
STUN is a very "ping like" (UDP) protocol for a server to echo back to a client what IP and port it sees the client as. The client can then use the response from the server and compare the result with what it thinks its locally enumerated IP address is. If the server's response matches the locally enumerated IP address, the client host can self determinte that it is directly connected to the Internet. Otherwise, the client must assume it is behind a NAT - but for the majority of routers, you have just created a port mapping that can be used for other P2P connection scenarios.
Further, you can you use the RESPONSE-PORT attribute in the STUN binding request for the server to respond back to a different port. This will effectively allow you to detect if you are firewalled or not.
TCP - this gets a little tricky. STUN can partially be used to determine if you are behind a NAT. Or simply making an http request to whatismyip.com and parsing the result to see if there's a NAT. But it gets tricky, as there's no service on the internet that I know of that will test a TCP connection back to you.
With all the above in mind, the vast majority of broadband users are likely behind a NAT that also acts as a firewall. Either given by their ISP or their own wireless router device. And even if they are not, most operating systems have some sort of minimal firewall to block unsolicited traffic. So it's very limiting to have a P2P client out there than can only work on direct connections.
With that said, on Windows (and likely others), you can program your app's install package can register with the Windows firewall so your it is not blocked. But if you aren't targeting Windows, you may have to ask the user to manually fix his firewall software.
Oh shameless plug. You can use this open source STUN server and client library which supports all of the semantics described above. Follow up with me offline if you need access to a stun service.
You might find this article useful
http://msdn.microsoft.com/en-us/library/aa364726%28v=VS.85%29.aspx
I would start with each os and ask if firewall services are turned on. Secondly, I would attempt the socket connections and determine from the error codes if connections are being reset or timeout. I'm only familiar with winsock coding, so I can't really say much for Linux or mac os.
I'm developing a SIP mobile softphone, customer needs a complete hiding of SIP messages from softphones to SIP servers as VOIP calls are regionally prohibited, however using TLS connection was not sufficient since the message headers are easily recognized as a SIP message. What are the best common alternative?
what about openvpn, IPSec tunneling?
Transmitting SIP over TLS means the SIP headers will only be viewable if someone is able to compromise your TLS keys, i.e. it's highly unlikely unless some national security agency is on your case.
What you might be encountering is port 5061 being blocked since it's the default and therefore well known SIP TLS port. To get around that simply use a different port for your SIP TLS connection. As far as anyone viewing the traffic goes if it's not suing port 5061 they won't have any idea that SIP is being used in your TLS stream.
Of course you also need to consider the RTP traffic which is what will carry the audio part of the call once SIP has set it up. There are no standardised ports for RTP but some popular VoIP softswitches do use certain ranges by default. For example Asterisk uses UDP 10,000 to 20,000. To work around that you'd really need to use SRTP but that's going to be harder to set up since not that many SIP user agents and servers support it. It will also be easier to detect for someone watching your traffic since even without knowing the contents the profile of RTP packets would be detectable. Still it's likely to need a sophisticated entity monitoring your traffic to detect a VoIP call using SIP over TLS on a non-standard port and SRTP call amongst the general noise of internet traffic.
Such as Skype/Team viewer/Logmein etc application, which send audio/video behind NAT (behind firewall). But when i make a small tiny application which send text to another NAT location it failed to do the same.
Example:
Sender:
-> Public ip: 91.1.2.3 My lan ip is: 192.168.1.2 with port 14446 udp
-------> Data format: RTP packets
Receiver:
<------- Data received: 0 packets
-> Public ip: 92.1.2.3 Friend lan ip is: 10.0.0.2 with port 14446 udp
* same in both way
How others does this? What is the way of doing peer 2 peer application development to overcome NAT issues? Always we have public ip's and mostly it has NAT issues.
But how does then Skype works in such cases too? Do we have a audio/video port range for UDP or always UDP is open from anything? But mine does not work above range ports for UDP i also tried. What is the secret? that is making me curious!!.
Note:
My goal is audio packets handling where i believe too much filtering or firewall cause latency and delay and other issues gets involved relatively too. So i would like to know very clearly for my application that some of the ports (which port ranges?) can be used for such purposes, where it really not blocking development stress.
There are a number of types of NATs, which vary in what traffic they'll allow in.
See the Wikipedia article on NATs
For most NATs, STUN will let you open ports AND find out what port you opened (may be different than the port you sent from). In SIP and RTSP you'd typically provide the external IP and port determined by STUN to the other end.
A fully-symmetric NAT means that STUN won't let you use a 3rd-party server to prop ports via STUN, so you'll have to use UPnP (if enabled) or map ports in the router (or set up triggers), or you'll have to play evil games to make both sides think they initiated the connection. (Not easy and not guaranteed.)
See the ICE & TURN specs (RFCs) from the IETF for detailed mechanisms to traverse NATs - though note that in some cases you must use an external proxy to forward packets.
One common solution is that the client program connects outward to the server and thus establishes a connection. Most firewalls allow outward connections - the assumption being that you are trusted and can always connect to the outside. When the server then wishes to send a message to you, it responds on the open connection.
I believe the port that you use is what is usually used to determine if it should be allowed or not. Certain ports are always let through. I'm not sure of the exact ports, but that will be different for all NATs and firewalls.
How can an application be designed such that two peers can communicate directly with each other (assuming both know each other's IPs), but without outgoing connections? That's, no ports will be opened. Bitorrent for example does it, but multiplayer games (as far as I know) require port forwarding.
I'm not sure what you mean by No Outgoing Connections, I'm going to assume like everyone else you meant no Incoming Connections (they are behind a NAT/FW/etc).
The most common one mentioned so far is UPNP, which in this context is a protocol that allows you as a computer to talk to the Gateway and say forward me this port because I want someone on the outside to be able to talk to me. UPNP is also designed for other things, but this is the common thing for home networking (Actually it's one of many definitions).
There are also more common and slightly more reliable ways if you don't own the network. The most common is called STUN but if I recall correctly there are a few variants. Basically you use a third party server that allows incoming connections to try and coordinate a communication channel. Basically, what you do is send a UDP packet to you're peer, which will open up you're NAT for a response, but gets dropped on you're peer's NAT (since no forwarding rule exists yet). Through the connection to the intermediary, they are then told to do the same, which now opens up their NAT, and matches the existing rule in you're NAT. Now the communications can proceed. Their is a variant of this which will allow a TCP/IP connection as well by sending SYN and SYN-ACK messages with some coordination.
The Wikipedia articles I've linked to has links to the relevant rfc's for these protocols on precisely how they work. Essentially it comes down to, there isn't an easy answer, as this is a very network centric problem.
You need a "meeting point" in the network somewhere: the participants "meet" at a "gateway" of some sort and the said "gateway function" takes care of the forwarding.
At least that's one way of doing it: I won't try to comment on the details of Bittorrent... I am sure you can google for links.
UPNP dealt with this mostly in the recent years, but the need to open ports is because the application has been coded to listen on a specific port for a response.
Ports beneath 1024 are called "registered" because they've been assigned a port number because a company paid for it. This doesn't mean you couldn't use port 53 for a webserver or SSH, just that most will assume when they see it that they are dealing with DNS. Ports above 1024 are unregistered, so there's no association - your web browser, be it Internet Explorer/Firefox/etc, is using an unregistered port to send the request to the StackOverflow webserver(s) on port 80. You can use:
netstat -a
..on windows hosts to see what network connections are currently established, including the port involved.
UPNP can be used to negotiate with the router to open and forward a port to your application. Even bit-torrent needs at least one of the peers to have an open port to enable p2p connections. There is no need for both peers to have an open port however, since they both communicate with the same server (tracker) that lets them negotiate and determine who has an open port.
An alternative is an echo-server / relay-server somewhere on the internet that both peers trust, and have that relay all the traffic.
The "problem" with this solution is that the echo-server needs to have lots of bandwidth to accomodate all connected peers since it relays all the traffic rather than establish p2p connections.
Check out EchoWare: http://www.echogent.com/tech.htm