Unable to register to FreeSwitch server & unable to call SIP client (XLite) respectively using SIPml5 client

StackOverflow https://stackoverflow.com/questions/20517496

  •  31-08-2022
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Question

I am unable to register to FreeSwitch server & unable to call to SIP client (XLite) by using SIPml5 SIP client.

Following is my HTML5 code:

<!DOCTYPE html>
<html>

<head>

<meta content="charset=utf-8"/>
<script type="text/javascript" src="SIPml-api.js"></script>
<title>SIP Client 1</title>

<script type="text/javascript">
window.onload = function()
{
var readyCallback = function(e){
    createSipStack(); // see next section
};
    var errorCallback = function(e){
    console.error('Failed to initialize the engine: ' + e.message);
}
SIPml.init(readyCallback, errorCallback);

console.info("WINDOW ONLOAD");
}
       var sipStack;
        function createSipStack()
        {
            var eventsListener = function(e)
            {
                if(e.type == 'started')
                {
                    login();
                }
                else if(e.type == 'i_new_message')
                {
                    acceptMessage(e);
                }
                else if(e.type == 'i_new_call')
                {
                    acceptCall(e);
                }
            }
            sipStack = new SIPml.Stack(
            {
                realm: '192.168.1.33', // mandatory: domain name
                impi: '1001', // mandatory: authorization name (IMS Private Identity)
                impu: 'sip:1001@192.168.1.33', // mandatory: valid SIP Uri (IMS Public Identity)
                password: '1234',
                display_name: 'Osama', // optional
                enable_rtcweb_breaker: true,
                events_listener: { events: '*', listener: eventsListener } // optional: '*' means all events
            });
            sipStack.start();
            console.info("CREATE SIP STACK");

        }

        var registerSession;
        var eventsListener = function(e){
            console.info('session event = ' + e.type);
            if(e.type == 'connected' && e.session == registerSession){
                makeCall();
            }
        }
        function login(){
            registerSession = sipStack.newSession('register', {
                events_listener: { events: '*', listener: eventsListener } // optional: '*' means all events
            });
            registerSession.register();
            console.info("LOGIN");
        }


        var callSession;
        function makeCall()
        {
            var eventsListener = function(e)
            {
                console.info('session event = ' + e.type);
            }
            callSession = sipStack.newSession('call-audiovideo', {
                video_local: document.getElementById('video_local'),
                video_remote: document.getElementById('video_remote'),
                audio_remote: document.getElementById('audio_remote'),
                events_listener: { events: '*', listener: eventsListener } // optional: '*' means all events
            });
            callSession.call('1002');
            console.info("MAKE CALL");
        }

        function acceptCall(e){
            e.newSession.accept(); // e.newSession.reject() to reject the call
        }

</script>
</head>
<body>

<form method="post" style="height: 197px">
<input name="Button1" type="button" value="Call" onclick="makeCall()">&nbsp;
<input name="Button2" type="button" value="Accept" onclick="acceptCall"><video class="video" id="video_remote" autoplay="autoplay" style="width: 184px; height: 188px"></video>&nbsp;
<audio id="audio_remote" autoplay="autoplay" />
</audio><video class="video" width="88px" height="72px" id="video_local" autoplay="autoplay"></video><span></span></form>

</body>

</html>

Following is my Chrome console output:

SIPML5 API version = 1.3.203 SIPml-api.js:1
User-Agent=Mozilla/5.0 (Windows NT 6.1; WOW64) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/31.0.1650.63 Safari/537.36 SIPml-api.js:1
WebSocket supported = yes SIPml-api.js:1
Navigator friendly name = chrome SIPml-api.js:1
OS friendly name = windows SIPml-api.js:1
Have WebRTC = yes SIPml-api.js:1
Have GUM = yes SIPml-api.js:1
Engine initialized SIPml-api.js:1
s_websocket_server_url=(null) SIPml-api.js:1
s_sip_outboundproxy_url=(null) SIPml-api.js:1
b_rtcweb_breaker_enabled=yes SIPml-api.js:1
b_click2call_enabled=no SIPml-api.js:1
b_early_ims=yes SIPml-api.js:1
b_enable_media_stream_cache=no SIPml-api.js:1
o_bandwidth={} SIPml-api.js:1
o_video_size={} SIPml-api.js:1
SIP stack start: proxy='ns313841.ovh.net:10062', realm='<sip:192.168.1.33>', impi='1001', impu='"Osama"<sip:1001@192.168.1.33>' SIPml-api.js:1
Connecting to 'wss://ns313841.ovh.net:10062' SIPml-api.js:1
CREATE SIP STACK default.html:52
WINDOW ONLOAD default.html:21
__tsip_transport_ws_onopen SIPml-api.js:1
State machine: tsip_dialog_register_Started_2_InProgress_X_oRegister SIPml-api.js:1
SEND: REGISTER sip:192.168.1.33 SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKpMrcuIswwvJBNixNUiHLuVPJA7tXxnrN;rport
From: "Osama"<sip:1001@192.168.1.33>;tag=UAy54NJ9TiBQ4J8hWEn8
To: "Osama"<sip:1001@192.168.1.33>
Contact: "Osama"<sips:1001@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=wss>;expires=1800;click2call=no
Call-ID: 690d263e-325b-1fc8-5bf7-bc5a545940f1
CSeq: 2210 REGISTER
Content-Length: 0
Max-Forwards: 70
Supported: path

SIPml-api.js:1
LOGIN default.html:68
session event = connecting default.html:58
session event = sent_request default.html:58
State machine: tsip_dialog_register_Any_2_Terminated_X_transportError SIPml-api.js:1
=== REGISTER Dialog terminated === SIPml-api.js:1
session event = transport_error default.html:58
session event = terminated default.html:58
The FSM is in the final state

I followed this developer guide. I don't know where am going wrong. I even enabled RTCWEB breaker as mentioned here. Please help me in registering to FreeSwitch server & calling to SIP client using SIPml5 client. The XLite is registered to FreeSwitch & is in ready state.

Was it helpful?

Solution

You need to install webrtc2sip in order to work with web client. I also faced the same problem i.e. transport error but after installing webrtc2sip this problem has been resolved. I am able to connect to FreeSwitch successfully although now there are other problems.

to install it Go to this link - http://code.google.com/p/webrtc2sip/wiki/Building_Source_v2_0

it looks lengthy but its straight forward.

Also don' forget to read the technical guide here.

After successfull installation, run this command in terminal 

sudo netstat -nlpa | grep webrtc

and find the socket address of webrtc2sip server which should look like "192.168.1.2:10060" and put this address in your code in front of websocket address. ws://192.168.1.2:10060 precisely.

OTHER TIPS

you have too set the ws(web socket) like that : websocket_proxy_url: 'ws://192.168.200.10:8088/ws'

I had a look into sipml source code and seems that transport implementation expects to receive a domain in realm variable but you directly set an IP address. Have you tried to set realm to "freeswitch.org"?

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