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@lists.freeswitch.org [mailto:freeswitch-users-bounces@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@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.
>>
>> -Dan