[Freeswitch-users] webrtc client using sip.js and conference api interaction

[Giovanni Maruzzelli]

On Wed, May 27, 2020 at 12:04 PM Jonathan Hunter <jhunter at voxboxcoms.co.uk>
wrote:

>
> Does anyone know best practice here to be able to interact with the
> freeswitch conference API so I can mute users and list participants from a
> web browser, ideally without using dtmf?
>
> I have previously used Verto, but due to scale I am using
> kamailio/rtpengine as a websockets gateway which then load balances
> requests to freeswitch so its only dealing with SIP and RTP.
>
> All works fine, I just want to now be able to kick users/list participants
> and so on via the browser so wondered the best way to do this now I am now
> using verto, as I used to subscribe to live array for that but now I am
> using sip.js to fire SIP requests over websockets, how do I send conference
> api commands and to the correct box as well if a farm of freeswitch servers
> are in use.
>


You cannot.

I use SIP MESSAGE for conveying this kind of commands and statuses, I had
to write all the logic (client side and server side)

I will probably made opensource my SIP video conference client for
FreeSWITCH in next months, but nothing certain yet.

Have a nice continuation,

-giovanni





>
>
> Many thanks
>
> Jon
>
>
>
>
>
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-- 
Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
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