Hello, I'm trying to set up TLS-protected FS, and cannot resolve TLS+NAT incoming calls issue. My clients are CSipSimple on Android and XLite on PC. Everything works fine when calling to non-TLS client, activating TLS yields (CSipSimple -> XLite): 2013-01-14 05:08:46.941041 [DEBUG] sofia_glue.c:3648 sofia/internal/1001@sip.over.ru Set rtp dtmf delay to 40 2013-01-14 05:08:46.941041 [NOTICE] sofia_glue.c:4259 Pre-Answer sofia/internal/1001@sip.over.ru! 2013-01-14 05:08:46.941041 [DEBUG] switch_channel.c:3136 (sofia/internal/1001@sip.over.ru) Callstate Change RINGING -> EARLY 2013-01-14 05:08:46.941041 [DEBUG] mod_sofia.c:856 Local SDP sofia/internal/1001@sip.over.ru: v=0 o=FreeSWITCH 1358119830 1358119831 IN IP4 88.85.71.230 s=FreeSWITCH c=IN IP4 88.85.71.230 t=0 0 m=audio 20296 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2013-01-14 05:08:46.941041 [DEBUG] switch_core_session.c:830 Send signal sofia/internal/1001@sip.over.ru [BREAK] 2013-01-14 05:08:46.941041 [DEBUG] switch_channel.c:3395 (sofia/internal/1001@sip.over.ru) Callstate Change EARLY -> ACTIVE 2013-01-14 05:08:46.941041 [NOTICE] mod_dptools.c:1176 Channel [sofia/internal/1001@sip.over.ru] has been answered EXECUTE sofia/internal/1001@sip.over.ru sleep(1000) 2013-01-14 05:08:46.941041 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/1001@sip.over.ru [BREAK] 2013-01-14 05:08:46.941041 [DEBUG] sofia.c:5599 Channel sofia/internal/1001@sip.over.ru entering state [completed][200] 2013-01-14 05:08:46.981041 [DEBUG] switch_rtp.c:928 [ zrtp utils]: Send <HELLO> ssrc=1379485558 seq=19624 size=140. Stream 48:CLEAR:START 2013-01-14 05:08:47.101068 [DEBUG] switch_rtp.c:928 [ zrtp utils]: Send <HELLO> ssrc=1379485558 seq=19625 size=140. Stream 48:CLEAR:START 2013-01-14 05:08:47.121057 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/1001@sip.over.ru [BREAK] 2013-01-14 05:08:47.121057 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/1001@sip.over.ru [BREAK] 2013-01-14 05:08:47.121057 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/1001@sip.over.ru [BREAK] 2013-01-14 05:08:47.141041 [DEBUG] sofia.c:5599 Channel sofia/internal/1001@sip.over.ru entering state [ready][200] 2013-01-14 05:08:47.281041 [INFO] switch_rtp.c:3642 Auto Changing port from 95.27.78.36:8081 to 95.27.78.36:42659 2013-01-14 05:08:47.301066 [DEBUG] switch_rtp.c:928 [ zrtp utils]: Send <HELLO> ssrc=1379485558 seq=19626 size=140. Stream 48:CLEAR:START 2013-01-14 05:08:47.501043 [DEBUG] switch_rtp.c:928 [ zrtp utils]: Send <HELLO> ssrc=1379485558 seq=19627 size=140. Stream 48:CLEAR:START 2013-01-14 05:08:47.701074 [DEBUG] switch_rtp.c:928 [ zrtp engine]: WARNING! HELLO have been resent 5 times without a response. Raising ZRTP_EVENT_NO_ZRTP_QUICK event. ID=48 2013-01-14 05:08:47.701074 [DEBUG] switch_rtp.c:928 [ zrtp utils]: Send <HELLO> ssrc=1379485558 seq=19628 size=140. Stream 48:CLEAR:START 2013-01-14 05:08:47.901047 [DEBUG] switch_rtp.c:928 [ zrtp utils]: Send <HELLO> ssrc=1379485558 seq=19629 size=140. Stream 48:CLEAR:START EXECUTE sofia/internal/1001@sip.over.ru bridge(loopback/app=voicemail:default 88.85.71.230 1000) Calling from XLite to CSIpSimple is somehow different, after about a minute of conversation it breaks, but with the same "HELLO have been resent 5 times" in logs. It is possible at all to interconnect two NATed clients using SIP/TLS? If yes, what softphones proved working and how should I setup FS/phones? Alex