[Freeswitch-users] SIP/TLS, INVITE and NAT

[Alex Povolotsky]

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 at 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 at sip.over.ru!
2013-01-14 05:08:46.941041 [DEBUG] switch_channel.c:3136 
(sofia/internal/1001 at sip.over.ru) Callstate Change RINGING -> EARLY
2013-01-14 05:08:46.941041 [DEBUG] mod_sofia.c:856 Local SDP 
sofia/internal/1001 at 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 at sip.over.ru [BREAK]
2013-01-14 05:08:46.941041 [DEBUG] switch_channel.c:3395 
(sofia/internal/1001 at sip.over.ru) Callstate Change EARLY -> ACTIVE
2013-01-14 05:08:46.941041 [NOTICE] mod_dptools.c:1176 Channel 
[sofia/internal/1001 at sip.over.ru] has been answered
EXECUTE sofia/internal/1001 at sip.over.ru sleep(1000)
2013-01-14 05:08:46.941041 [DEBUG] switch_core_session.c:975 Send signal 
sofia/internal/1001 at sip.over.ru [BREAK]
2013-01-14 05:08:46.941041 [DEBUG] sofia.c:5599 Channel 
sofia/internal/1001 at 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 at sip.over.ru [BREAK]
2013-01-14 05:08:47.121057 [DEBUG] switch_core_session.c:975 Send signal 
sofia/internal/1001 at sip.over.ru [BREAK]
2013-01-14 05:08:47.121057 [DEBUG] switch_core_session.c:975 Send signal 
sofia/internal/1001 at sip.over.ru [BREAK]
2013-01-14 05:08:47.141041 [DEBUG] sofia.c:5599 Channel 
sofia/internal/1001 at 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 at 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