<div dir="ltr"><div class="gmail_default" style="font-family:courier new,monospace">Hi guys!<br><br></div><div class="gmail_default" style="font-family:courier new,monospace">I´ve been tying to test Freeswitch 1.5 (latest git) for WebRTC, but so far I have been very unsuccessful.<br><br></div><div class="gmail_default" style="font-family:courier new,monospace">I have a server, connected directly to internet, NO NAT on server side.<br><br><br></div><div class="gmail_default" style="font-family:courier new,monospace">FS ----> INTERNET <--- NAT ---- CLIENTS<br><br></div><div class="gmail_default" style="font-family:courier new,monospace">My vars.conf includes this:<br><br><br></div><div class="gmail_default" style="font-family:courier new,monospace"> <X-PRE-PROCESS cmd="set" data="bind_server_ip=xxx.xxx.xxx.xxx"/><br> <X-PRE-PROCESS cmd="set" data="external_rtp_ip=xxx.xxx.xxx.xxx"/><br> <X-PRE-PROCESS cmd="set" data="external_sip_ip=xxxx.xxx.xxx.xxx"/><br><br> <!-- Internal SIP Profile --><br> <X-PRE-PROCESS cmd="set" data="internal_auth_calls=true"/><br> <X-PRE-PROCESS cmd="set" data="internal_sip_port=5060"/><br> <X-PRE-PROCESS cmd="set" data="internal_tls_port=5061"/><br> <X-PRE-PROCESS cmd="set" data="internal_ssl_enable=true"/><br> <X-PRE-PROCESS cmd="set" data="internal_ssl_dir=/opt/CloudVoice-vPBX/fs-20150506/certs/"/><br><br> <!-- External SIP Profile --><br> <X-PRE-PROCESS cmd="set" data="external_auth_calls=false"/><br> <X-PRE-PROCESS cmd="set" data="external_sip_port=5080"/><br> <X-PRE-PROCESS cmd="set" data="external_tls_port=5081"/><br> <X-PRE-PROCESS cmd="set" data="external_ssl_enable=true"/><br><br><br></div><div class="gmail_default" style="font-family:courier new,monospace">On my external profile I have this relevants lines...<br><br> <param name="aggressive-nat-detection" value="true"/><br> <param name="apply-inbound-acl" value="domains"/><br> <param name="local-network-acl" value="localnet.auto"/><br> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/><br> <param name="ext-sip-ip" value="$${external_sip_ip}"/><br> <param name="ws-binding" value=":5066"/><br> <param name="wss-binding" value=":7443"/><br><br><br></div><div class="gmail_default" style="font-family:courier new,monospace">When doing some testing....<br><br></div><div class="gmail_default" style="font-family:courier new,monospace">Calling to echo test <br><br> 2015-05-07 20:51:14.124955 [NOTICE] switch_channel.c:1075 New Channel sofia/internal/<a href="mailto:1007@webrtc.cibersys.com">1007@webrtc.cibersys.com</a> [39b777d1-6e61-49ec-bfd0-6f7ca0bd0caf]<br>2015-05-07 20:51:14.384990 [INFO] mod_dialplan_xml.c:635 Processing test <1007>->9196 in context default<br>2015-05-07 20:51:14.384990 [WARNING] switch_core_media.c:2791 NO candidate ACL defined, Defaulting to wan.auto<br>2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2829 Save audio Candidate cid: 1 proto: UDP type: host addr: <a href="http://10.0.0.126:57630">10.0.0.126:57630</a><br>2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2829 Save audio Candidate cid: 1 proto: UDP type: host addr: <a href="http://192.168.56.1:57631">192.168.56.1:57631</a><br>2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2829 Save audio Candidate cid: 2 proto: UDP type: host addr: <a href="http://10.0.0.126:57632">10.0.0.126:57632</a><br>2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2829 Save audio Candidate cid: 2 proto: UDP type: host addr: <a href="http://192.168.56.1:57633">192.168.56.1:57633</a><br>2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2824 Choose audio Candidate cid: 1 proto: UDP type: srflx addr: <a href="http://201.210.31.83:57630">201.210.31.83:57630</a><br>2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2824 Choose audio Candidate cid: 2 proto: UDP type: srflx addr: <a href="http://201.210.31.83:57632">201.210.31.83:57632</a><br>2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2953 setting remote audio ice addr to <a href="http://201.210.31.83:57630">201.210.31.83:57630</a> based on candidate<br>2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2978 setting remote rtcp audio addr to <a href="http://201.210.31.83:57632">201.210.31.83:57632</a> based on candidate<br>2015-05-07 20:51:14.384990 [INFO] switch_core_media.c:5345 Activating Audio ICE<br>2015-05-07 20:51:14.384990 [NOTICE] switch_rtp.c:4019 Activating RTP audio ICE: 76ffb71e:TduhdjNUFKq3Vzaz <a href="http://201.210.31.83:57630">201.210.31.83:57630</a><br>2015-05-07 20:51:14.384990 [INFO] switch_core_media.c:5388 Activating RTCP PORT 57632<br>2015-05-07 20:51:14.384990 [INFO] switch_core_media.c:5398 Activating RTCP ICE<br>2015-05-07 20:51:14.384990 [NOTICE] switch_rtp.c:4019 Activating RTCP audio ICE: 76ffb71e:TduhdjNUFKq3Vzaz <a href="http://201.210.31.83:57632">201.210.31.83:57632</a><br>2015-05-07 20:51:14.384990 [INFO] switch_rtp.c:3103 Activate RTP/RTCP audio DTLS client<br>2015-05-07 20:51:14.384990 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/<a href="mailto:1007@webrtc.cibersys.com">1007@webrtc.cibersys.com</a>!<br>2015-05-07 20:51:14.384990 [NOTICE] mod_dptools.c:1292 Channel [sofia/internal/<a href="mailto:1007@webrtc.cibersys.com">1007@webrtc.cibersys.com</a>] has been answered<br>2015-05-07 20:51:14.864993 [INFO] switch_rtp.c:2924 Changing audio DTLS state from HANDSHAKE to SETUP<br>2015-05-07 20:51:14.864993 [INFO] switch_rtp.c:2832 audio Fingerprint Verified.<br>2015-05-07 20:51:14.864993 [INFO] switch_rtp.c:3384 Activating Audio Secure RTP SEND<br>2015-05-07 20:51:14.864993 [INFO] switch_rtp.c:3362 Activating Audio Secure RTP RECV<br>2015-05-07 20:51:14.864993 [INFO] switch_rtp.c:2872 Changing audio DTLS state from SETUP to READY<br>2015-05-07 20:52:15.564950 [INFO] switch_core_media.c:3039 Activating audio RTCP PORT 57632<br>2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2829 Save audio Candidate cid: 1 proto: UDP type: host addr: <a href="http://10.0.0.126:57630">10.0.0.126:57630</a><br>2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2829 Save audio Candidate cid: 1 proto: UDP type: host addr: <a href="http://192.168.56.1:57631">192.168.56.1:57631</a><br>2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2829 Save audio Candidate cid: 2 proto: UDP type: host addr: <a href="http://10.0.0.126:57632">10.0.0.126:57632</a><br>2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2829 Save audio Candidate cid: 2 proto: UDP type: host addr: <a href="http://192.168.56.1:57633">192.168.56.1:57633</a><br>2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2824 Choose audio Candidate cid: 1 proto: UDP type: srflx addr: <a href="http://201.210.31.83:57630">201.210.31.83:57630</a><br>2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2824 Choose audio Candidate cid: 2 proto: UDP type: srflx addr: <a href="http://201.210.31.83:57632">201.210.31.83:57632</a><br>2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2953 setting remote audio ice addr to <a href="http://201.210.31.83:57630">201.210.31.83:57630</a> based on candidate<br>2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2978 setting remote rtcp audio addr to <a href="http://201.210.31.83:57632">201.210.31.83:57632</a> based on candidate<br>2015-05-07 20:52:15.564950 [INFO] switch_core_media.c:2995 RE-Activating audio ICE<br>2015-05-07 20:52:15.564950 [NOTICE] switch_rtp.c:4019 Activating RTP audio ICE: 76ffb71e:TduhdjNUFKq3Vzaz <a href="http://201.210.31.83:57630">201.210.31.83:57630</a><br>2015-05-07 20:52:15.564950 [INFO] switch_core_media.c:3039 Activating audio RTCP PORT 57632<br>2015-05-07 20:52:15.564950 [INFO] switch_core_media.c:3050 Activating audio RTCP ICE<br>2015-05-07 20:52:15.564950 [NOTICE] switch_rtp.c:4019 Activating RTCP audio ICE: 76ffb71e:TduhdjNUFKq3Vzaz <a href="http://201.210.31.83:57632">201.210.31.83:57632</a><br>2015-05-07 20:52:15.564950 [INFO] switch_core_media.c:5030 RE-SETTING video DTLS<br>2015-05-07 20:52:15.704955 [WARNING] switch_rtp.c:878 sofia/internal/<a href="mailto:1007@webrtc.cibersys.com">1007@webrtc.cibersys.com</a> got stun binding response 487 Role Conflict<br>2015-05-07 20:52:15.704955 [WARNING] switch_rtp.c:889 Changing role to CONTROLLED<br><br></div><div class="gmail_default" style="font-family:courier new,monospace">No audio<br><br></div><div class="gmail_default" style="font-family:courier new,monospace">When calling another ext... no audio en the webrtc side.<br><br></div><div class="gmail_default" style="font-family:courier new,monospace">Can somebody help me by pointing out the right direction?<br><br></div><div class="gmail_default" style="font-family:courier new,monospace">Ive been using FF and Chrome with sipML5<br></div><div class="gmail_default" style="font-family:courier new,monospace"><br></div><div class="gmail_default" style="font-family:courier new,monospace"><br clear="all"></div><br>-- <br><div class="gmail_signature"><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><font size="2"><span style="font-family:courier new,monospace"><br><img src="https://www.cibersys.com/imagenes/logotipo-cibersys-the-new-easy.png"><br><br>Víctor E. Medina M.<br></span></font><div><font size="2"><span style="font-family:courier new,monospace">Platform Architect / Chief Infrastructure<br></span></font></div><font size="2"><span style="font-family:courier new,monospace"><span style="display:inline"><span style="display:inline"><a>+58424 291 4561</a></span></span><br>BB #79A8AFA2<br>@VMCibersys<br></span></font></div><div dir="ltr"><font size="2"><span style="font-family:courier new,monospace"><br></span></font></div></div></div></div></div></div></div></div>
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