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

[Michael Jerris]

why would you set sip_secure_media?

On Jun 26, 2013, at 10:41 AM, Gerald Weber <gerald.weber at besharp.at> wrote:

> Hi,
>  
> how did you get audio working ?
> I’ve also tried jssip and latest chrome and sip works, but there is no audio. Codecs like opus,   ilbc and isac are installed.
> In the dialplan i set sip_secure_media=true but the speaker stays silent. I’ve tried ws and wss bindings but no luck.
>  
> Thanks for any suggestions,
> gw
>  
> Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von João Mesquita
> Gesendet: Mittwoch, 26. Juni 2013 15:28
> An: FreeSWITCH Users Help
> Betreff: Re: [Freeswitch-users] FreeSWITCH adds WebRTC support to new 1.4 BETA.
>  
> 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.
> >>
> >> -Dan
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