[Freeswitch-users] Digging into WebRTC for the first time

Tue Sep 20 12:24:23 MSD 2016

Here is jthe simple example taht includes sip.js just to dial in to FS:

<html>
<style>
video {
    border: 1px solid lightgrey;
}
</style>
<body>

<video id="localVideo" muted="muted"></video>
<video id="remoteVideo"></video>
<br>
<!--button id="startCall">Start Call</button-->
<button id="endCall">End Call</button>

<script src="sip-0.7.5.min.js"></script> //download this sip.js version and
place it in same dir as this html file or include from public link

<script id="e">
//var my_session;

var startButton = document.getElementById('startCall');

var my_session = document.addEventListener('DOMContentLoaded', function() {
my_session = userAgent.invite('sip:dial_string at your.dns.name', options);
}, false);

var endButton = document.getElementById('endCall');

endButton.addEventListener("click", function () {
    my_session.bye();
}, false);

var options = {
    media: {
        constraints: {
            audio: true,
            video: true
        },
        render: {
            local: document.getElementById('localVideo'),
            remote: document.getElementById('remoteVideo')
        }
    }
};

var userAgent = new SIP.UA({
 traceSip: true,
 uri: 'dial_string at your.dns.name',
 wsServers: ['wss://your.dns.name:port'], //wss port of verto or sofia
 authorizationUser: 'directory_user',
 password: 'pass',
 hackIpInContact: true,
 hackWssInTransport: true,
 stunServers: ["stun.l.google.com:19302","stun1.l.google.com:19302","
stun2.l.google.com:19302","stun3.l.google.com:19302","
stun4.l.google.com:19302"]
});

</script>

</body>
</html>

so just google for sip-0.7.5.min.js download it or replace it with public
link.
Make sure to replace all dial_string, pass, your.dns.name:port values that
are for your server.
If you have proper certificates, for wss to be shaked on :) it will work, i
suggest letsencrypt

Good luck :)
Mirko

On Tue, Sep 20, 2016 at 1:47 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> A lot of it is covered in https://freeswitch.org/
> confluence/display/FREESWITCH/Debian+8+Jessie though it goes a lot
> further and gives you a working video conference over webrtc.
>
> Loot at https://freeswitch.org/confluence/display/FREESWITCH/mod_verto as
> well.
>
> Also maybe look at http://sipjs.com/
>
>
> On Mon, Sep 19, 2016 at 6:43 PM, Don Hawkins <hawkins at hawkinsegroup.com>
> wrote:
>
>> I'm trying to get WebRTC going on our FS server for the first time. FS is
>> not my problem though, WebRTC is.
>>
>> All I need is a one button script that dials into FS, at which point I
>> will put the user in a conference, anyone have any example scripts that
>> already do this?
>>
>> I've looked at all the free libraries and unfortunately java is not my
>> thing so I've had no success.
>>
>> Thanks!
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
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>>
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>>
>
>
>
> --
> Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬
>
> ☞ http://freeswitch.org/  ☞ http://cluecon.com/  ☞
> http://twitter.com/FreeSWITCH
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> <http://freeswitch.org/g+>*
>
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>
> https://www.youtube.com/watch?v=9XXgW34t40s
> https://www.youtube.com/watch?v=NLaDpGQuZDA
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
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-- 
Regards,
Mirko

¯\_(ツ)_/¯
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