Hi Giovanni,
I'm not sure now since call set up works in cantina.
I'm using standard FS setup with 7443 port for wss and simple (50 lines) of sip.js code, which I can share. Using google stun servers.
So this setup still works with older versions.
ICE candidates are saved normally on new version, but then after:
2016-07-27 09:43:11.092485 [INFO] switch_core_media.c:7059 Activating Video ICE
I get:
2016-07-27 09:43:11.092485 [ALERT] switch_rtp.c:7171 Skip sending video packet 0 bytes (ice not ready @ line 4366!)
and after Callstate Change RINGING -> EARLY it fails on both audio and video port:
2016-07-27 09:43:11.092485 [ALERT] switch_rtp.c:7171 Skip sending video packet 0 bytes (ice not ready @ line 4366!)
2016-07-27 09:43:11.512489 [ERR] switch_rtp.c:3112 video Handshake failure 1
2016-07-27 09:43:11.512489 [INFO] switch_rtp.c:3113 Changing video DTLS state from HANDSHAKE to FAIL
2016-07-27 09:43:11.542511 [ERR] switch_rtp.c:3112 audio Handshake failure 1
2016-07-27 09:43:11.542511 [INFO] switch_rtp.c:3113 Changing audio DTLS state from HANDSHAKE to FAIL
Maybe the google stun server is failing to send initial rtp.
Thanks,
mirko