Greetings,
i have been trying to create an web app that connects with an webrtc client (jssip, sipml5 or sip.js) to my freepbx 14, all of them give the same result to Mozilla/5.0, even back tracked to chrome 49 and have the same issues.
Audio= works perfect both ways.
Video= softphone or hardphone receives video but browser wont show video.
dtmf= works both ways.
i tested jssip, sipml5, sip.js clients demos and all gave the same problem.
i tried this kurento asterisk tutorial from webrtc ventures (cant post links), but it crashes whenever i pick up the phone with this TypeError: this.mediaHandler.hasDescription is not a function.
every information about webrtc clients is atleast 5 months old so im not even sure is it possible to achieve.
im not forced to use freepbx 14, i could revert to asterisk 11, but still dont know if that would change anything.
this is my webrtc extension:
[2000] ;kurento-appserver
host=dynamic
secret=asterikpwd
context=from-internal
transport=ws,wss,udp
type=friend
encryption=no
avpf=yes
icesupport=yes
directmedia=no
disallow=all
allow=ullaw,opus
allow=vp8
videosupport=yes
im using chan_sip cause pjsip throws forbidden 403