I've set up a Kamailio v4.0 SIP registrar and proxy for SIP over websockets, and everything is working fine except sending outbound messages or making outbound calls to a SIP address on a foreign domain. I can receive messages and calls from non-local SIP addresses, but not send any out.
I've tried to understand the documentation of various modules like domain and outbound and I'm still stumped. My configuration file is essentially identical to this one.
This feels less like configuration and more like learning a new programming language. Any ideas on how to proceed?
edit: spotted the following in the log file:
INFO: rr [../outbound/api.h:49]: Failed to import bind_ob
INFO: rr [rr_mod.c:159]: outbound module not available
Recompiled with STUN=1 and loaded outbound.so in config (is this barking up the wrong tree?). Now log says:
INFO: rr [rr_mod.c:156]: Bound rr module to outbound module
[...]
WARNING: <core> [msg_translator.c:2499]: TCP/TLS connection (id: 0) for WebSocket could not be found
ERROR: <core> [msg_translator.c:1725]: could not create Via header
ERROR: tm [t_fwd.c:435]: could not build request
The logs suggest that no connection to the WebSocket client can be found. Kamailio cannot open a connection to a WebSocket (webrtc) softphone, if one does not exist, the SIP message cannot be delivered.
You can list active TCP connections using kamcmd tool:
kamcmd core.tcp_list
If you use WebSockets over TLS:
kamcmd tls.list
There is a config function that should be used in this case - set_forward_no_connect() - to avoid getting so far in processing.
Related
I'm building a message broker which communicates with clients over ZeroMQ PUSH/PULL sockets and has the ability to exclude clients from messages they're not subscribed to from the server side (unlike ZeroMQ pub/sub which excludes messages on the client side).
Currently, I implement it in the following way:
Server: Binds ZeroMQ PULL socket on a fixed port
Client: Binds a ZeroMQ PULL socket on a random or fixed port
Client: Connects to the server's PULL socket and sends a handshake message containing the new client's address and port.
Server: Recieves handshake from client and connects a PUSH socket to the client's PULL server. Sends handshake response to the client's socket.
Client: Recieves handshake. Connected!
Now the client and server can communicate bidirectionally and the server can send messages to only a certain subset of clients. It works great!
However, this model doesn't work if the clients binding PULL sockets are unable to open a port in their firewall so the server can connect to them. How can I resolve this with minimal re-architecting (as the current model works very well when the firewall can be configured correctly)
I've considered the following:
Router/dealer pattern? I'm fairly ignorant on this and documentation I found was sparse.
Some sort of transport bridging? The linked example provides an example for PUB/SUB.
I was hoping to get some advice from someone who knows more about ZeroMQ than me.
tl;dr: I implemented a message broker that communicates with clients via bidirectional push/pull sockets. Each client binds a PULL socket and the server keeps a map of PUSH sockets so that it can address specific subscribers. How do I deal with a firewall blocking the client ports?
You can use the router/dealer to do this like you say. By default the ROUTER socket tracks every connection it has. The way it does this is by having the caller stick the connection identity information in front of each message it recieves. This makes things like pub/sub fairly trivial as all you need to do is handle a few messages server side that the DEALER socket sends it. In the past I have done something like
1.) Server side is a ROUTER socket. The ROUTER handles 2 messages from DEALER sockets SUB/UNSUB. This alongside the identity info sent as the first part of a frame allows the router to know the messages that a client is interested in.
2.) The server checks the mapping to see which clients should be sent a particular type of data using the map and then forwards the message to the correct client by appending the identity again to the start of the message.
This is nice in that it allows a single port to be exposed on the server. Client side we do not need to expose ports, simply just connect to the server ROUTER socket.
See https://zguide.zeromq.org/docs/chapter3/ for more info.
I am working on a product which uses ZeroMQ (version 4.0.1).
The server and client communicate based on ZeroMQ ROUTER-socket.
To read socket events, server and client also create socket-monitor sockets (PAIR). There are three ports on which server binds and listens. Out of these three ports, one port is in a non-secured mode. Other two ports are using md5-authentication.
The issue I am facing is that, both the server and the client spontaneously receive socket disconnect for one of the secure port sockets (please see a log below). I have checked multiple times that server and client both have L3 reachability to each other.
What else I should check for?
What really triggers this error scenario?
zmq_print_callback:ZmQ: int zmq::stream_engine_t::read(void*, size_t):923
Stream engine recv():
TCP socket (187) to unknown:0 was disconnected
with error 107 [Transport endpoint is not connected]
Below sequence of events can trigger this error on server
Server receives ACCEPTED event for clientY and gets FD1.
Link-flap/network issue happens and clientY disconnects but server does not receive this disconnect.
Network recovers and clientY connects back to server.
Server receives ACCEPTED event for clientY and gets FD2. However, packets sent to this sockets does not go out of the server.
After 1 min or so, clientY receives "Transport endpoint is not connected error" for FD1.
Application can use this to treat as client disconnect.
Given a URL that ends with .svc and that is supposed to run a SOAP web service, how can I get some data from it?
I tried:
to access it via a web browser
to access it via the Python's library Zeep
to access it via the Microsoft utilitary svcutil.exe
In all cases, I get a timeout error:
Failed to establish a new connection: [Errno 10060] A connection attempt failed because the connected party did not properly respond after a period of time.
Does it mean that the web service does not work, or that I do things the wrong way?
Importantly - there is a big distinction between "service not active" (and by that I mean no listener on port 80), and "port not open in firewall".
If the problem were simply that you didn't have a service listening on port 80, you would have gotten something like "connection reset" or "connection rejected" as an error.
Instead, you appear to have gotten a timeout, which implies that either the SYN from the client doesn't reach the server, or the SYN/ACK from the server doesn't reach the client. [ You could verify this by doing a packet capture for port 80 on both client and server ]
I would be tempted to check any firewall in front of the server to see that it's letting port 80 traffic through from your client.
Diagnosing Connectivity Issues
Without more details it is difficult to say, but given your timeout error:
Failed to establish a new connection: [Errno 10060] A connection attempt failed because the connected party did not properly respond after a period of time.
This indicates a network connectivity error at the TCP level, so it is likely web service is not active on the port your are using (default of 80 for http, 443 for https).
In a comment you said you pinged the URL and it responded normally - I assume this means you pinged the hostname. If this is responding normally it means the server is active, but that doesn't tell you anything about the availability of the web service on that server.
telnet %hostname% %port%
where %port% is 80 for http or 443 for https, or something else if there is a port number in the URL you are using (e.g. http://somehost.somewhere.com:port/path.scv)
If ping works and telnet does not connect, then the service is not active.
I suspect this is the case. If the service was active and it was simply that you requesting the data incorrectly, I believe you'd get a different error message - e.g. a valid HTTP response with status code 500 or 404 or similar.
Getting Data from a Web Service
As to your original question as to how to get data from it - once you verify that the service is active, the method to get the data will depend on the specification of the service - i.e.:
which HTTP methods (GET, POST, etc.) does it support
what parameters it requires
what format it requires the parameters in
are the parameters in the query string or POST body.
To interact with a web service there are many command line tools that can be used, as well as the options you have tried, including:
POSTMan Google Chrome Plugin
curl
wget
In windows Powershell, the Invoke-WebRequest
Getting Data from a SOAP Web Service
As you have said it is a SOAP web service, if you have the URL for the wsdl, you can often interract with it using Powershell SOAP WebService Proxies.
The wsdl location varies, but is often at a URL that looks something like.
http://host/path.svc?wsdl
http://host/path.svc/?wsdl
http://host/path/?wsdl
Also if it's configured correctly, just loading the URL in a browser will present a page with a link to the wsdl.
The general idea is:
$URI="http://hostname/path.svc?wsdl"
$Proxy = New-WebserviceProxy $URI –Namespace X
$Proxy | get-member -MemberType Method
This will return a list of methods on the proxy that you can invoke as powershell methods. Any types defined in the wsdl that are needed for arguments, or returned from methods will be available within the namespace X. Invoking the methods will proxy the request to the service, taking care of serializing parameters and serializing results into powershell objects.
I have 2 sip clients on the same computer.
Both of them is registering to a server that is running on port 5060.
For the first client the UDP is on port 5060 and for the other is 5061. When I come from one client to another, after the ringing part i receive the error:
only one usage of each socket address is normally permited.
Got any ideas why I got this error?
Your server and client are both trying to use port 5060, hence the error message. Change the first client to use 5062 or something else.
Also, 5061 is normally used for secured SIP (normal listening port + 1 in the proxy/server). Do not use it for the second client.
It means you're clients are both trying to claim the same socket for the communication channel, or the server is trying to reclaim the socket given to client A, to reuse it for client B.
The software handeling the socket, should be smart enough to rely on the OS to assign port numbers instead of hardcoding the port numbers in the code, this is a 100% guarantee for socket issues.
I currently have issues with SIP User Agents behind a symmetric NAT connecting to my SIP client, which is an IVR voice service.
I read that Asterisk has a solution for this in the sip.conf, where I can set attribute
nat=yes
and this will ignore the IP and Port in the SIP headers and use the one for the SIP request and also waits for an incoming RTP stream to reply to.
I'd like to make use of this feature as we already have an Asterix server installed for AIX requests.
What would be the minimum configuration required for Asterix to act as the man in the middle on a new port as 5060 will still be used to connect directly to the SIP client? I don't care about authentication etc. I just need the Asterix to act as a SIP relay.
Thanks
K
You may also need to have "canreinvite=no" in the [general] section of your sip.conf.
That setting keeps Asterisk in the call path, otherwise voice traffic may be sent directly from one endpoint to the other.