I was thinking mostly on separating roles. Think of a site that would have a click to call feature that has webrtc on it. You would want to have that profile completely secured and isolated from any other connection, right? Regards João Mesquita FreeSWITCH™ Solutions On Wed, Jun 26, 2013 at 11:28 AM, Michael Jerris <mike@jerris.com> wrote:
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?
On Jun 26, 2013, at 9:28 AM, João Mesquita <jmesquita@freeswitch.org> wrote:
Whooray!!!! I've installed the new beta yesterday and tried it out with Chrome+JsSIP and it works fantastic!!!
One question tho, is it possible/desirable to have a sofia profile to work only with webrtc and not with udp or tcp bindings?
I haven't tried and I guess I should've before asking dumb questions but....
João Mesquita FreeSWITCH™ Solutions
On Tue, Jun 25, 2013 at 7:42 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
Now also working in latest FireFox 22
sipml5 works with both chrome and FF
https://webrtc.freeswitch.org/sipml5/
On Tue, Jun 25, 2013 at 2:04 AM, Michael Jerris <mike@jerris.com> wrote:
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.
Mike
On Jun 25, 2013, at 12:37 AM, Jeff Leung <jleung@v10networks.ca> wrote:
Media wise, how would that be handled with web sockets?
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch- users-bounces@lists.freeswitch.org] On Behalf Of Dan Barua Sent: Monday, June 24, 2013 3:10 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] FreeSWITCH adds WebRTC support to new 1.4 BETA.
I asked the same question on Google+, Michael Jerris helpfully replied:
you just need to setup FreeSWITCH as normal, and add 2 configuration params for the binding addresses for the websockets and secure websockets. the params are ws-binding and wss-binding format is ip:port or just :port if your using the same ip as sip-ip
I haven't got round to trying it myself yet.
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