+1<div><br></div><div>We already have profiles binding to both <span></span>UDP and TCP - why not WS too? :)<div><br></div><div>-Steve</div><div><br><br>On Wednesday, June 26, 2013, Michael Jerris wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div style="word-wrap:break-word">I suspect if you do transport param like transport=ws it would work but I haven't tried that. I'm not sure why it would be desirable because you can't create outbound websocket connections, so why would you want it on a different profile?<div>
<br><div><div>On Jun 26, 2013, at 9:28 AM, Joćo Mesquita <<a href="javascript:_e({}, 'cvml', 'jmesquita@freeswitch.org');" target="_blank">jmesquita@freeswitch.org</a>> wrote:</div><br><blockquote type="cite">
<div dir="ltr">Whooray!!!! I've installed the new beta yesterday and tried it out with Chrome+JsSIP and it works fantastic!!!<div><br></div><div>One question tho, is it possible/desirable to have a sofia profile to work only with webrtc and not with udp or tcp bindings?</div>
<div><br></div><div>I haven't tried and I guess I should've before asking dumb questions but....</div></div><div class="gmail_extra"><br clear="all"><div>Joćo Mesquita<br>FreeSWITCH Solutions<br></div>
<br><br><div class="gmail_quote">On Tue, Jun 25, 2013 at 7:42 PM, Anthony Minessale <span dir="ltr"><<a href="javascript:_e({}, 'cvml', 'anthony.minessale@gmail.com');" target="_blank">anthony.minessale@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
<div dir="ltr">Now also working in latest FireFox 22 <div><br></div><div>sipml5 works with both chrome and FF</div><div><br></div><div><a href="https://webrtc.freeswitch.org/sipml5/" target="_blank">https://webrtc.freeswitch.org/sipml5/</a><br>
</div><div><br></div></div><div class="gmail_extra"><div><div><br><br><div class="gmail_quote">On Tue, Jun 25, 2013 at 2:04 AM, Michael Jerris <span dir="ltr"><<a href="javascript:_e({}, 'cvml', 'mike@jerris.com');" target="_blank">mike@jerris.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">Media is not handled by websockets. Web-sockets are a new transport for sip signaling. If an endpoint registers from a websocket transport, when we send it a call, we automatically send it an sdp appropriate for webrtc. If you want to send a call with an sdp for webrtc that isn't for a registration over websockets, just add media_webrtc=true var to the origination vars. If we receive an sdp from webrtc, we will automatically handle it.<br>
<br>
Mike<br>
<div><br>
On Jun 25, 2013, at 12:37 AM, Jeff Leung <<a href="javascript:_e({}, 'cvml', 'jleung@v10networks.ca');" target="_blank">jleung@v10networks.ca</a>> wrote:<br>
<br>
> Media wise, how would that be handled with web sockets?<br>
><br>
>> -----Original Message-----<br>
>> From: <a href="javascript:_e({}, 'cvml', 'freeswitch-users-bounces@lists.freeswitch.org');" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a> [mailto:<a href="javascript:_e({}, 'cvml', 'freeswitch-');" target="_blank">freeswitch-</a><br>
>> <a href="javascript:_e({}, 'cvml', 'users-bounces@lists.freeswitch.org');" target="_blank">users-bounces@lists.freeswitch.org</a>] On Behalf Of Dan Barua<br>
>> Sent: Monday, June 24, 2013 3:10 PM<br>
>> To: <a href="javascript:_e({}, 'cvml', 'freeswitch-users@lists.freeswitch.org');" target="_blank">freeswitch-users@lists.freeswitch.org</a><br>
>> Subject: Re: [Freeswitch-users] FreeSWITCH adds WebRTC support to new<br>
>> 1.4 BETA.<br>
>><br>
>> I asked the same question on Google+, Michael Jerris helpfully replied:<br>
>><br>
>> you just need to setup FreeSWITCH as normal, and add 2 configuration<br>
>> params for the binding addresses for the websockets and secure<br>
>> websockets. the params are ws-binding and wss-binding format is ip:port or<br>
>> just :port if your using the same ip as sip-ip<br>
>><br>
>><br>
>> I haven't got round to trying it myself yet.<br>
>><br>
></div></blockquote></div></div></div></div></blockquote></div></div></blockquote></div></div></div></blockquote></div></div>