Hi All, I use FreeSwitch for transcoding SRTP from WebRTC to SIP with RTP and vice versa. When Firefox version 38 add requires for Perfect Forward Secrecy (PFS) I have problem with Handshake in FS when Firefox Browser terminate a call. I found that that problem was fixed in the version 1.4.19 https://freeswitch.org/jira/browse/FS-7425 I have the same problem in the version 1.4.20 eff635a6-33a0-11e5-ac73-891cb7dea826 2015-07-26 14:16:46.564182 [INFO] switch_rtp.c:2924 Changing audio DTLS state from HANDSHAKE to SETUP eff635a6-33a0-11e5-ac73-891cb7dea826 2015-07-26 14:16:46.564182 [INFO] switch_rtp.c:2832 audio Fingerprint Verified. eff635a6-33a0-11e5-ac73-891cb7dea826 2015-07-26 14:16:46.564182 [INFO] switch_rtp.c:3384 Activating Audio Secure RTP SEND eff635a6-33a0-11e5-ac73-891cb7dea826 2015-07-26 14:16:46.564182 [INFO] switch_rtp.c:3362 Activating Audio Secure RTP RECV eff635a6-33a0-11e5-ac73-891cb7dea826 2015-07-26 14:16:46.564182 [INFO] switch_rtp.c:2872 Changing audio DTLS state from SETUP to READY eff635a6-33a0-11e5-ac73-891cb7dea826 2015-07-26 14:16:46.604193 [DEBUG] switch_rtp.c:1937 rtcp_stats_init: ssrc[-1374535478] base_seq[6179] f01c46d8-33a0-11e5-ac83-891cb7dea826 2015-07-26 14:16:46.644210 [DEBUG] switch_rtp.c:5884 Correct ip/port confirmed. f02184f4-33a0-11e5-ac91-891cb7dea826 2015-07-26 14:16:47.264283 [ERR] switch_rtp.c:2917 audio Handshake failure 1 f02184f4-33a0-11e5-ac91-891cb7dea826 2015-07-26 14:16:47.264283 [INFO] switch_rtp.c:2918 Changing audio DTLS state from HANDSHAKE to FAIL f02184f4-33a0-11e5-ac91-891cb7dea826 2015-07-26 14:16:47.264283 [NOTICE] switch_rtp.c:2899 Hangup sofia/external/992555500004@voip.webrtc.jajah.com [CS_EXCHANGE_MEDIA] [DESTINATION_OUT_OF_ORDER] what additional definition I should make to resolve this problem? Thanks, Alex Polischuk