webrtc -Click2call with asterisk server , Without user registration ( No Call_ID )

[KaRuNaKaR Reddy]

hey guys, 

 my plan is to develop a same like application "click2call " , But is it possible to do it with out user registration, i can see that  if i can just  send the invite message with sdp (With out call-ID) , can we able to make a connection and rtp path between end points, 

I do not need any incoming calls to my mobile browser , i just want to place the outbound call,

thanks in advance

[Mamadou]

http://click2dial.org/ meet your requirements:
- Open Source and free
- No need to register. When you press the call button we just send a
SIP INVITE.
- Works with Asterisk. On the website there is a "call us" button
(green, on the right side) and when you press it you will hear a voice
mail from Asterisk if no one answer the call.
- Works with any SIP provider (sip2sip.info, iptel.org...)
- Works on Chrome, Firefox Nightly and Bowser without additional
plugins
For any question, you can ask on our dev-group: http://groups.google.com/group/doubango

[Warren McDonald]

Hi,

I think the point is to not require registration with a sip server. There are several examples of SIP soft clients that can call to properly addressed endpoints without requiring registration to a SIP server. This does obviously not support calling to the soft client, only outbound. 

So the question may be - can a call be made without requiring a SIP registrar to be specified.

Warren

[Anders]

That is the case.

I'm part of the team that Karunakar works with. What we are doing is trying to develop our own Click2Call service, making calls without register ourselves. 

We have a SIPML5 and Asterisk environment setup and it works (see screenshot), but our values are "hard coded". Now we want to abstract away the "Connect" and "Disconnect" functionality and let through calls just by pushing a button and therefore route the call to a given number.

Is that possible?

[Bossiel]

Yes. This is already done.

Sent from my iPhone

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[Mamadou]

Your problem is that you want to start from the SIPML5 softphone which includes too many features (registration, incoming calls, hold/resume, transfer, multi-line,...) to have something very basic. Send an INVITE without registration is as simple as:

SIPml.init(
                                function(e){
                                    var stack =  new SIPml.Stack({realm: 'example.org', impi: 'bob', impu: 'sip:b...@example.org', password: 'mysecret',
                                        events_listener: { events: 'started', listener: function(e){
                                                    var callSession = stack.newSession('call-audiovideo', {
                                                            audio_remote: document.getElementById('audio-remote')
                                                        });
                                                    callSession.call('alice');
                                                } 
                                            }
                                    });
                                    stack.start();
                                }
                        );

Off course you've to change it to not create a stack for each call. Check the API at http://sipml5.org/docgen/symbols/SIPml.html

PS: You should send questions related to SIPML5 or webrtc2sip to doubango group: https://groups.google.com/forum/#!forum/doubango

[Mamadou]

oops... replace 'call-audiovideo' with 'call-audio' as there is no <video /> elements.