<html><head><meta http-equiv="Content-Type" content="text/html; charset=utf-8"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; line-break: after-white-space;" class="">FS won’t but a client that supports turn could use that via the turn server.<div class=""><br class=""><div><br class=""><blockquote type="cite" class=""><div class="">On Sep 3, 2020, at 10:07 PM, David P <<a href="mailto:davidswalkabout@gmail.com" class="">davidswalkabout@gmail.com</a>> wrote:</div><br class="Apple-interchange-newline"><div class=""><div dir="ltr" class=""><div dir="ltr" class="">Hi Brian W,<div class=""><br class=""></div><div class="">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:</div><div class=""><a href="http://lists.freeswitch.org/pipermail/freeswitch-users/2015-September/115727.html" class="">http://lists.freeswitch.org/pipermail/freeswitch-users/2015-September/115727.html</a><br class=""></div><div class=""><br class=""></div><div class="">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?</div><div class=""><br class=""></div><div class=""><span style="color:rgb(23,43,77);font-family:SFMono-Medium,"SF Mono","Segoe UI Mono","Roboto Mono","Ubuntu Mono",Menlo,Consolas,Courier,monospace;font-size:12px;white-space:pre;background-color:rgb(244,245,247)" class=""><action application="conference_set_auto_outcall"
data="['conference_member_flags=endconf,jitterbuffer_msec=5p:100p,media_webrtc=true']sofia/gateway/{hidden}"/></span><br class=""></div><div class=""></div></div><div class=""><br class=""></div></div></div></blockquote></div><br class=""></div></body></html>