[Freeswitch-users] Handle media of all calls via port 443?

[Mike Jerris]

FS won’t but a client that supports turn could use that via the turn server.


> On Sep 3, 2020, at 10:07 PM, David P <davidswalkabout at gmail.com> wrote:
> 
> Hi Brian W,
> 
> Back on Aug 10, Sergey Safarov suggested we might get FS's answer sdp to include more candidates via variable "media_webrtc=true". (We'd like TCP via port 443.) The only documentation I can find is an old post by you:
> http://lists.freeswitch.org/pipermail/freeswitch-users/2015-September/115727.html <http://lists.freeswitch.org/pipermail/freeswitch-users/2015-September/115727.html>
> 
> It didn't work for me when I set "media_webrtc" as a regular dialplan var, but your post mentions the {} of a bridge. We use a conference, so should we set it this way?
> 
> <action application="conference_set_auto_outcall"
>         data="['conference_member_flags=endconf,jitterbuffer_msec=5p:100p,media_webrtc=true']sofia/gateway/{hidden}"/>
> 

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