<div dir="ltr">Please check <b><a href="https://freeswitch.org/jira/browse/FS-7839">https://freeswitch.org/jira/browse/FS-7839</a> </b><br></div><div class="gmail_extra"><br><div class="gmail_quote">On Mon, Jul 27, 2015 at 8:33 AM, Alex Polischuk <span dir="ltr"><<a href="mailto:alxpol@gmail.com" target="_blank">alxpol@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div>Hi All,</div><div><br></div><div>I use FreeSwitch for transcoding SRTP from WebRTC to SIP with RTP and vice versa.</div><div>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 <a href="https://freeswitch.org/jira/browse/FS-7425" target="_blank">https://freeswitch.org/jira/browse/FS-7425</a></div><div><br></div><div>I have the same problem in the version 1.4.20</div><div><br></div><div>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</div><div>eff635a6-33a0-11e5-ac73-891cb7dea826 2015-07-26 14:16:46.564182 [INFO] switch_rtp.c:2832 audio Fingerprint Verified.</div><div>eff635a6-33a0-11e5-ac73-891cb7dea826 2015-07-26 14:16:46.564182 [INFO] switch_rtp.c:3384 Activating Audio Secure RTP SEND</div><div>eff635a6-33a0-11e5-ac73-891cb7dea826 2015-07-26 14:16:46.564182 [INFO] switch_rtp.c:3362 Activating Audio Secure RTP RECV</div><div>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</div><div>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]</div><div>f01c46d8-33a0-11e5-ac83-891cb7dea826 2015-07-26 14:16:46.644210 [DEBUG] switch_rtp.c:5884 Correct ip/port confirmed.</div><div>f02184f4-33a0-11e5-ac91-891cb7dea826 2015-07-26 14:16:47.264283 [ERR] switch_rtp.c:2917 audio Handshake failure 1</div><div>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</div><div>f02184f4-33a0-11e5-ac91-891cb7dea826 2015-07-26 14:16:47.264283 [NOTICE] switch_rtp.c:2899 Hangup sofia/external/<a href="mailto:992555500004@voip.webrtc.jajah.com" target="_blank">992555500004@voip.webrtc.jajah.com</a> [CS_EXCHANGE_MEDIA] [DESTINATION_OUT_OF_ORDER]</div><div><br></div><div><br></div><div>what additional definition I should make to resolve this problem?</div><div><br></div><div>Thanks,</div><div>Alex Polischuk</div><div><br></div></div>
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