Kamailio / Freeswitch connection with rooms

[Kerugal]

Hi Aaron Clauson, we are currently trying the following UseCase: A user is supposed to connect to a Kamailo/Freeswitch on an iOS Unity client via WebRTC + SIP. The Freeswitch is working with the following descriptor. For Example:

SIPCallDescriptor callDescriptor = new SIPCallDescriptor(
            currentFSInfo.username,                                         // username
            currentFSInfo.password,                                         // password
            $"sip:{currentRoomNumber.ToString()}@{currentFSInfo.hostname}", // uri
            $"sip:{currentFSInfo.username}@{currentFSInfo.hostname}",       // from
            $"sip:{currentRoomNumber.ToString()}@{currentFSInfo.hostname}", // to   => room id 
            null, null, null,                                               // routeset, customHeaders, authUsername
            SIPCallDirection.Out,                                           // call direction
            "application/sdp",                                              // content type
            currentModifiedOffer,                                           // content
            null);

Unfortunately, the SIPClientUserAgent does not take the correct paramters that are needed. Should look something like this: new SIPClientUserAgent(sipTransport, invalidUser, invalidHost, sipInstanceId, currentRoomNumber.ToString()); Is such a scenario known, or is the library not designed for it at all?

By the way - Thanks for the great work

[Kerugal]

Small addendum:

Furthermore, we currently connect via JSSip in our WebApp. Unfortunately I don't see a way to work with sockets in the registration - like: https://jssip.net/documentation/3.2.x/api/ua_configuration_parameters/#parameter_sockets

[sipsorcery]

WebRTC+SIP from Unity on iOS??!! That's very cool. If you have any demoes/examples/anything I'd love to see that in action.

Easy one first. The SIP over websockets should be doable. Take a look at this example. The sipsorcery library does support websockets as a transport protocol.

The SIPCallDescriptor can almost certainly take the parameters you need to place a call to FreeSWITCH. It's problem is being too flexible and therefore unweildy to use. Is there any reason you're using SIPClientUserAgent instead of SIPUserAgent? The SIPClientUserAgent is a lower level primitive and will take more wiring up. Whereabouts are you wiring up WebRTC in that scenario? The Asterisk ICE example may be a useful starting point. It's establishing a SIP call with Asterisk and using a WebRTC RTCPeerConnection for the media. FreeSWITCH should work with the example as is.

[Kerugal]

Hello Aaron Clauson, Ok, after I unfortunately couldn't manage to establish a connection in different ways, I'll get back to you. The last example you gave: SIPWebSocketChannel as well as SIPClientWebSocketChannel unfortunately I could not establish a connection.

var webSocketChannel = new SIPClientWebSocketChannel();
sipTransport.CanCreateMissingChannels = false;
sipTransport.AddSIPChannel(webSocketChannel);

sipRegistrationUserAgent = 
new SIPRegistrationUserAgent(sipTransport, USERNAME, PASSWORD, DESTINATION, EXPIRY);

Basically, I am looking for a way to establish a connection like JSSIP, via the WebSocket interface: https://jssip.net/documentation/2.0.x/api/websocket/

var socket = new JsSIP.WebSocketInterface('wss://foo.bar.net');

    var configuration = {
      sockets: [socket],
      uri: 'sip:foo_user@foo.bar.net',
      password: '123456654321'
    };
    ua = new JsSIP.UA(configuration);
    ua.start();

Sorry if I am missing obvious things. But I am confused how a WebSocket connection can be passed through the existing classes? Especially about the default to pass an IPEndPoint or IPAddress. Or am I using SIPClientWebsocketChannel incorrectly? How exactly should the ServerUri be entered there?

Thanks for your work Patrick

Small addendum: the SIPClientWebSocket pulls its URL from the RIPRegistrationsUserAgent hostname. In our case, however, the WS-URI is not the same as the SIP-URI. WS-URI: wss://foo.bar.net/sip SIP-URI: sip:foo.bar.net (without the /sip)

[sipsorcery]

If you want to specify a web socket transport on your SIP URI you can use:

sip:user@localhost:80;transport=ws

And for a TLS web socket:

sip:user@localhost:443;transport=wss

A handy testing tool is the sipcmdline utility:

examples\sipcmdline>dotnet run -- -d sip:user@localhost:443;transport=wss -s reg -v