Hi! I am trying to test Freeswitch WebRTC capabiities. We are using JsSIP try web site (http://tryit.jssip.net/) Our servers are located on Amazon EC2 so, there is a NAT firewall between them. Freeswitch version is: FreeSWITCH Version 1.5.15b+git~20150404T093935Z~6eb59c7119~64bit (git 6eb59c7 2015-04-04 09:39:35Z 64bit) - TODAY MASTER. When calling to echoetst ext. 9196, and using the "video/audio" call it works perfectly, when calling without the Video enable call is placed but no audio is heard. The same thing happens when calling to the ext 3000 ext conference. When using the video enabled everything seems to work just fine. But if I disable video tha call wont work. Can someone try to help in here? My relevant vars.xml lines: <X-PRE-PROCESS cmd="set" data="domain=ve001.voice.cibersys.com"/> <X-PRE-PROCESS cmd="set" data="domain_name=ve001.voice.cibersys.com"/> <X-PRE-PROCESS cmd="set" data="hold_music=local_stream://moh"/> <X-PRE-PROCESS cmd="set" data="use_profile=external"/> <X-PRE-PROCESS cmd="set" data="rtp_secure_media=optional:AES_CM_128_HMAC_SHA1_32"/> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=OPUS,PCMA,PCMU,G729,G722,H264,VP8,H263,H263-1998,H263-2000"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=OPUS,PCMA,PCMU,G729,G722,H264,VP8,H263,H263-1998,H263-2000"/> <X-PRE-PROCESS cmd="set" data="bind_server_ip=54.186.52.106"/> <X-PRE-PROCESS cmd="set" data="external_rtp_ip=54.186.52.106"/> <X-PRE-PROCESS cmd="set" data="external_sip_ip=54.186.52.106"/> <X-PRE-PROCESS cmd="set" data="sip_tls_version=tlsv1,tlsv1.1,tlsv1.2"/> <X-PRE-PROCESS cmd="set" data="sip_tls_ciphers=ALL:!ADH:!LOW:!EXP:!MD5:@STRENGTH"/> <!-- Internal SIP Profile --> <X-PRE-PROCESS cmd="set" data="internal_auth_calls=true"/> <X-PRE-PROCESS cmd="set" data="internal_sip_port=5090"/> <X-PRE-PROCESS cmd="set" data="internal_tls_port=5091"/> <X-PRE-PROCESS cmd="set" data="internal_ssl_enable=true"/> <X-PRE-PROCESS cmd="set" data="internal_ssl_dir=/opt/fs-20150407/certs/"/> <!-- External SIP Profile --> <X-PRE-PROCESS cmd="set" data="external_auth_calls=false"/> <X-PRE-PROCESS cmd="set" data="external_sip_port=5070"/> <X-PRE-PROCESS cmd="set" data="external_tls_port=5071"/> <X-PRE-PROCESS cmd="set" data="external_ssl_enable=false"/> My internal_profile <param name="liberal-dtmf" value="true"/> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="aggressive-nat-detection" value="true"/> <param name="multiple-registrations" value="true"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="NDLB-received-in-nat-reg-contact" value="true"/> <param name="NDLB-force-rport" value="true"/> <param name="NDLB-broken-auth-hash" value="true"/> <param name="enable-timer" value="false"/> <param name="auth-calls" value="true"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="wss-binding" value=":7443"/> Here is the log of the call to 9196 without audio and video (no video option selected) http://pastebin.com/AncwpiPx <- this doesn't work Here is the log of the call to 9196 with audio and video option selected http://pastebin.com/FT7XMpzU <- in this one audio and video works -- Víctor E. Medina M. Platform Architect / Chief Infrastructure +58424 291 4561 BB #79A8AFA2 @VMCibersys