<html><head><meta http-equiv="Content-Type" content="text/html charset=windows-1252"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;">Hello,<div><br></div><div>according to the instructions on <a href="https://freeswitch.org/confluence/display/FREESWITCH/WebRTC">https://freeswitch.org/confluence/display/FREESWITCH/WebRTC</a> I have enabled webrtc on one Sip profile. I’ve added my own cerificate and installed in /usr/local/freeswitch/certs. I’m connecting to freeswitch using JsSip library. The signaling part works fine, I can see all SIP messages in the log, but then it fails to establish the audio/rtp connection:</div><div><br></div><div><div style="margin: 0px; font-size: 14px; font-family: Menlo; color: rgb(52, 189, 38); background-color: rgb(0, 0, 0); position: static; z-index: auto;">2014-11-10 17:36:27.065688 [INFO] switch_core_media.c:5206 Skipping RTCP ICE (Same as RTP)</div><div style="margin: 0px; font-size: 14px; font-family: Menlo; color: rgb(52, 189, 38); background-color: rgb(0, 0, 0); position: static; z-index: auto;">2014-11-10 17:36:27.065688 [INFO] switch_rtp.c:3065 Activate RTP/RTCP audio DTLS client</div><div style="margin: 0px; font-size: 14px; font-family: Menlo; color: rgb(195, 55, 32); background-color: rgb(0, 0, 0); position: static; z-index: auto;">2014-11-10 17:36:27.065688 [ERR] switch_rtp.c:3117 audio DTLS key err [1]</div></div><div><br></div><div><br></div><div>I’m a bit confused about this message, I thought all tls settings in a sip profile and the dtls-srtp.pem certificate is not relevant for webrtc / wss, or what does this error message want to tell me?</div><div><br></div><div>Thanks</div><div><br></div><div>Markus </div></body></html>