Hi guys!

I´ve been tying to test Freeswitch 1.5 (latest git) for WebRTC, but so far I have been very unsuccessful.

I have a server, connected directly to internet, NO NAT on server side.

FS ----> INTERNET <--- NAT ---- CLIENTS

My vars.conf includes this:

  <X-PRE-PROCESS cmd="set" data="bind_server_ip=xxx.xxx.xxx.xxx"/>
  <X-PRE-PROCESS cmd="set" data="external_rtp_ip=xxx.xxx.xxx.xxx"/>
  <X-PRE-PROCESS cmd="set" data="external_sip_ip=xxxx.xxx.xxx.xxx"/>

  <!-- Internal SIP Profile -->
  <X-PRE-PROCESS cmd="set" data="internal_auth_calls=true"/>
  <X-PRE-PROCESS cmd="set" data="internal_sip_port=5060"/>
  <X-PRE-PROCESS cmd="set" data="internal_tls_port=5061"/>
  <X-PRE-PROCESS cmd="set" data="internal_ssl_enable=true"/>
  <X-PRE-PROCESS cmd="set" data="internal_ssl_dir=/opt/CloudVoice-vPBX/fs-20150506/certs/"/>

  <!-- External SIP Profile -->
  <X-PRE-PROCESS cmd="set" data="external_auth_calls=false"/>
  <X-PRE-PROCESS cmd="set" data="external_sip_port=5080"/>
  <X-PRE-PROCESS cmd="set" data="external_tls_port=5081"/>
  <X-PRE-PROCESS cmd="set" data="external_ssl_enable=true"/>

On my external profile I have this relevants lines...

 <param name="aggressive-nat-detection" value="true"/>
 <param name="apply-inbound-acl" value="domains"/>
 <param name="local-network-acl" value="localnet.auto"/>
 <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
 <param name="ext-sip-ip" value="$${external_sip_ip}"/>
 <param name="ws-binding"  value=":5066"/>
 <param name="wss-binding" value=":7443"/>

When doing some testing....

Calling to echo test

 2015-05-07 20:51:14.124955 [NOTICE] switch_channel.c:1075 New Channel sofia/internal/1007@webrtc.cibersys.com [39b777d1-6e61-49ec-bfd0-6f7ca0bd0caf]
2015-05-07 20:51:14.384990 [INFO] mod_dialplan_xml.c:635 Processing test <1007>->9196 in context default
2015-05-07 20:51:14.384990 [WARNING] switch_core_media.c:2791 NO candidate ACL defined, Defaulting to wan.auto
2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2829 Save audio Candidate cid: 1 proto: UDP type: host addr: 10.0.0.126:57630
2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2829 Save audio Candidate cid: 1 proto: UDP type: host addr: 192.168.56.1:57631
2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2829 Save audio Candidate cid: 2 proto: UDP type: host addr: 10.0.0.126:57632
2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2829 Save audio Candidate cid: 2 proto: UDP type: host addr: 192.168.56.1:57633
2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2824 Choose audio Candidate cid: 1 proto: UDP type: srflx addr: 201.210.31.83:57630
2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2824 Choose audio Candidate cid: 2 proto: UDP type: srflx addr: 201.210.31.83:57632
2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2953 setting remote audio ice addr to 201.210.31.83:57630 based on candidate
2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2978 setting remote rtcp audio addr to 201.210.31.83:57632 based on candidate
2015-05-07 20:51:14.384990 [INFO] switch_core_media.c:5345 Activating Audio ICE
2015-05-07 20:51:14.384990 [NOTICE] switch_rtp.c:4019 Activating RTP audio ICE: 76ffb71e:TduhdjNUFKq3Vzaz 201.210.31.83:57630
2015-05-07 20:51:14.384990 [INFO] switch_core_media.c:5388 Activating RTCP PORT 57632
2015-05-07 20:51:14.384990 [INFO] switch_core_media.c:5398 Activating RTCP ICE
2015-05-07 20:51:14.384990 [NOTICE] switch_rtp.c:4019 Activating RTCP audio ICE: 76ffb71e:TduhdjNUFKq3Vzaz 201.210.31.83:57632
2015-05-07 20:51:14.384990 [INFO] switch_rtp.c:3103 Activate RTP/RTCP audio DTLS client
2015-05-07 20:51:14.384990 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1007@webrtc.cibersys.com!
2015-05-07 20:51:14.384990 [NOTICE] mod_dptools.c:1292 Channel [sofia/internal/1007@webrtc.cibersys.com] has been answered
2015-05-07 20:51:14.864993 [INFO] switch_rtp.c:2924 Changing audio DTLS state from HANDSHAKE to SETUP
2015-05-07 20:51:14.864993 [INFO] switch_rtp.c:2832 audio Fingerprint Verified.
2015-05-07 20:51:14.864993 [INFO] switch_rtp.c:3384 Activating Audio Secure RTP SEND
2015-05-07 20:51:14.864993 [INFO] switch_rtp.c:3362 Activating Audio Secure RTP RECV
2015-05-07 20:51:14.864993 [INFO] switch_rtp.c:2872 Changing audio DTLS state from SETUP to READY
2015-05-07 20:52:15.564950 [INFO] switch_core_media.c:3039 Activating audio RTCP PORT 57632
2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2829 Save audio Candidate cid: 1 proto: UDP type: host addr: 10.0.0.126:57630
2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2829 Save audio Candidate cid: 1 proto: UDP type: host addr: 192.168.56.1:57631
2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2829 Save audio Candidate cid: 2 proto: UDP type: host addr: 10.0.0.126:57632
2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2829 Save audio Candidate cid: 2 proto: UDP type: host addr: 192.168.56.1:57633
2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2824 Choose audio Candidate cid: 1 proto: UDP type: srflx addr: 201.210.31.83:57630
2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2824 Choose audio Candidate cid: 2 proto: UDP type: srflx addr: 201.210.31.83:57632
2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2953 setting remote audio ice addr to 201.210.31.83:57630 based on candidate
2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2978 setting remote rtcp audio addr to 201.210.31.83:57632 based on candidate
2015-05-07 20:52:15.564950 [INFO] switch_core_media.c:2995 RE-Activating audio ICE
2015-05-07 20:52:15.564950 [NOTICE] switch_rtp.c:4019 Activating RTP audio ICE: 76ffb71e:TduhdjNUFKq3Vzaz 201.210.31.83:57630
2015-05-07 20:52:15.564950 [INFO] switch_core_media.c:3039 Activating audio RTCP PORT 57632
2015-05-07 20:52:15.564950 [INFO] switch_core_media.c:3050 Activating audio RTCP ICE
2015-05-07 20:52:15.564950 [NOTICE] switch_rtp.c:4019 Activating RTCP audio ICE: 76ffb71e:TduhdjNUFKq3Vzaz 201.210.31.83:57632
2015-05-07 20:52:15.564950 [INFO] switch_core_media.c:5030 RE-SETTING video DTLS
2015-05-07 20:52:15.704955 [WARNING] switch_rtp.c:878 sofia/internal/1007@webrtc.cibersys.com got stun binding response 487 Role Conflict
2015-05-07 20:52:15.704955 [WARNING] switch_rtp.c:889 Changing role to CONTROLLED

No audio

When calling another ext... no audio en the webrtc side.

Can somebody help me by pointing out the right direction?

Ive been using FF and Chrome with sipML5