[Freeswitch-users] FreeSWITCH adds WebRTC support to new 1.4 BETA.

[João Mesquita]

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 at 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 at 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 at 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 at 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 at v10networks.ca> wrote:
>>>
>>> > Media wise, how would that be handled with web sockets?
>>> >
>>> >> -----Original Message-----
>>> >> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>> freeswitch-
>>> >> users-bounces at lists.freeswitch.org] On Behalf Of Dan Barua
>>> >> Sent: Monday, June 24, 2013 3:10 PM
>>> >> To: freeswitch-users at 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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