[Freeswitch-users] Problem with TLS (call setup)

Krzysiek cris7 at o2.pl
Wed May 14 14:05:27 PDT 2008


Hi

I try to setup TLS SIP signalling between Freeswitch and Minisip softphone,
but it doesn't work.
I think I did everythnig ok in freeswitch configuration because 'sofia
profile' command printed that profile with tls is running on port 5061.
Minisip softphones registered properly(First I had to import a CA
certificate). Problem appears when I try to setup a call between them (the
same subnet). When one softphone try to setup a call, the second won't ring.
Here is a trace:
#######################################
freeswitch at trixswitch03> recv 669 bytes from tls/[192.168.1.3]:1072 at
20:33:52.506179:
   ------------------------------------------------------------------------
   INVITE sip:1001 at 192.168.1.2 SIP/2.0
   Via: SIP/2.0/TLS 192.168.1.3:1072;rport;branch=z9hG4bK28253
   Route: <sips:192.168.1.2:5061;lr=true;transport=tcp>
   From: <sip:1002 at 192.168.1.2>;tag=30769
   To: <sip:1001 at 192.168.1.2>
   Call-ID: 5206 at 192.168.1.3
   CSeq: 501 INVITE
   Max-Forwards: 70
   Contact: <sip:1002 at 192.168.1.3:1072;transport=tcp>;expires=1000
   User-Agent: Minisip
   Supported: 100rel,sdp-anat
   Content-Type: application/sdp
   Content-Length: 218

   v=0
   o=- 3344 3344 IN IP4 192.168.1.3
   s=Minisip Session
   t=0 0
   m=audio 32616 RTP/AVP 0 8 101
   c=IN IP4 192.168.1.3
   a=rtpmap:0 PCMU/8000/1
   a=rtpmap:8 PCMA/8000/1
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15
   ------------------------------------------------------------------------
send 268 bytes to tls/[192.168.1.3]:1072 at 20:33:52.509158:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/TLS 192.168.1.3:1072;rport=1072;branch=z9hG4bK28253
   From: <sip:1002 at 192.168.1.2>;tag=30769
   To: <sip:1001 at 192.168.1.2>
   Call-ID: 5206 at 192.168.1.3
   CSeq: 501 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.pre4-8086
   Content-Length: 0

   ------------------------------------------------------------------------
send 728 bytes to tls/[192.168.1.3]:1072 at 20:33:52.520721:
   ------------------------------------------------------------------------
   SIP/2.0 407 Proxy Authentication Required
   Via: SIP/2.0/TLS 192.168.1.3:1072;rport=1072;branch=z9hG4bK28253
   From: <sip:1002 at 192.168.1.2>;tag=30769
   To: <sip:1001 at 192.168.1.2>;tag=127yN23jBU4HF
   Call-ID: 5206 at 192.168.1.3
   CSeq: 501 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.pre4-8086
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: 100rel, precondition, timer
   Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary
   Proxy-Authenticate: Digest realm="192.168.1.2",
nonce="fef2ab8e-483f-45d6-8b13-59f383bd75fb", algorithm=MD5, qop="auth"
   Content-Length: 0

   ------------------------------------------------------------------------
recv 316 bytes from tls/[192.168.1.3]:1072 at 20:33:52.528439:
   ------------------------------------------------------------------------
   ACK sip:1001 at 192.168.1.2 SIP/2.0
   Via: SIP/2.0/TLS 192.168.1.3:1072;rport;branch=z9hG4bK28253
   Max-Forwards: 70
   Route: <sips:192.168.1.2:5061;lr=true;transport=tcp>
   From: <sip:1002 at 192.168.1.2>;tag=30769
   Call-ID: 5206 at 192.168.1.3
   CSeq: 501 ACK
   To: <sip:1001 at 192.168.1.2>;tag=127yN23jBU4HF
   Content-Length: 0

   ------------------------------------------------------------------------
recv 863 bytes from tls/[192.168.1.3]:1072 at 20:33:52.568375:
   ------------------------------------------------------------------------
   INVITE sip:1001 at 192.168.1.2 SIP/2.0
   Via: SIP/2.0/TLS 192.168.1.3:1072;rport;branch=z9hG4bK6868
   Route: <sips:192.168.1.2:5061;lr=true;transport=tcp>
   From: <sip:1002 at 192.168.1.2>;tag=30769
   To: <sip:1001 at 192.168.1.2>
   Call-ID: 5206 at 192.168.1.3
   CSeq: 502 INVITE
   Max-Forwards: 70
   Contact: <sip:1002 at 192.168.1.3:1072;transport=tcp>;expires=1000
   User-Agent: Minisip
   Supported: 100rel,sdp-anat
   Proxy-Authorization: Digest
algorithm=MD5,username="1002",realm="192.168.1.2",nonce="fef2ab8e-483f-45d6-8b13-59f383bd75fb",uri="sip:1001 at 192.168.1.2",response="cdf7016d5fe00d8c715047a9edc094e8"
   Content-Type: application/sdp
   Content-Length: 218

   v=0
   o=- 3344 3344 IN IP4 192.168.1.3
   s=Minisip Session
   t=0 0
   m=audio 32616 RTP/AVP 0 8 101
   c=IN IP4 192.168.1.3
   a=rtpmap:0 PCMU/8000/1
   a=rtpmap:8 PCMA/8000/1
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15
   ------------------------------------------------------------------------
send 267 bytes to tls/[192.168.1.3]:1072 at 20:33:52.570791:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/TLS 192.168.1.3:1072;rport=1072;branch=z9hG4bK6868
   From: <sip:1002 at 192.168.1.2>;tag=30769
   To: <sip:1001 at 192.168.1.2>
   Call-ID: 5206 at 192.168.1.3
   CSeq: 502 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.pre4-8086
   Content-Length: 0

   ------------------------------------------------------------------------
2008-05-14 22:33:52 [NOTICE] switch_channel.c:531 switch_channel_set_name()
New Channel sofia/default/1002 at 192.168.1.2
[025c4667-55eb-4db8-932a-c4117c810dd7]
2008-05-14 22:33:52 [INFO] mod_dialplan_xml.c:223 dialplan_hunt() Processing
1002->1001 at default
2008-05-14 22:33:52 [INFO] switch_ivr_async.c:1357
switch_ivr_bind_dtmf_meta_session() Bound: 1 execute_extension::dx XML
features
2008-05-14 22:33:52 [INFO] switch_ivr_async.c:1357
switch_ivr_bind_dtmf_meta_session() Bound: 2
record_session::/opt/freeswitch/recordings/1002.2008-05-14-22-33-52.wav
2008-05-14 22:33:52 [INFO] switch_ivr_async.c:1357
switch_ivr_bind_dtmf_meta_session() Bound: 3 execute_extension::cf XML
features
2008-05-14 22:33:52 [NOTICE] switch_channel.c:531 switch_channel_set_name()
New Channel sofia/default/1001 at 192.168.1.4:1065;transport=tcp
[3f4e1eba-46bd-4475-a3a6-bd37ce43821b]
#######################################

and few seconds later something like this appears:

#######################################


send 656 bytes to tls/[192.168.1.3]:1072 at 20:34:22.011100:
   ------------------------------------------------------------------------
   SIP/2.0 500 No session set by user
   Via: SIP/2.0/TLS 192.168.1.3:1072;rport=1072;branch=z9hG4bK6868
   From: <sip:1002 at 192.168.1.2>;tag=30769
   To: <sip:1001 at 192.168.1.2>;tag=2B1QQXmp83t4a
   Call-ID: 5206 at 192.168.1.3
   CSeq: 502 INVITE
   Contact: <sip:mod_sofia at 192.168.1.2:5060;transport=tcp>
   User-Agent: FreeSWITCH-mod_sofia/1.0.pre4-8086
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: 100rel, precondition, timer
   Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary
   Content-Length: 0

   ------------------------------------------------------------------------
recv 510 bytes from tls/[192.168.1.3]:1072 at 20:34:22.021547:
   ------------------------------------------------------------------------
   ACK sip:1001 at 192.168.1.2 SIP/2.0
   Via: SIP/2.0/TLS 192.168.1.3:1072;rport;branch=z9hG4bK6868
   Max-Forwards: 70
   Route: <sips:192.168.1.2:5061;lr=true;transport=tcp>
   From: <sip:1002 at 192.168.1.2>;tag=30769
   Call-ID: 5206 at 192.168.1.3
   CSeq: 502 ACK
   Proxy-Authorization: Digest
algorithm=MD5,username="1002",realm="192.168.1.2",nonce="fef2ab8e-483f-45d6-8b13-59f383bd75fb",uri="sip:1001 at 192.168.1.2",response="cdf7016d5fe00d8c715047a9edc094e8"
   To: <sip:1001 at 192.168.1.2>;tag=2B1QQXmp83t4a
   Content-Length: 0

   ------------------------------------------------------------------------
2008-05-14 22:34:22 [NOTICE] switch_ivr_originate.c:1154
switch_ivr_originate() Hangup
sofia/default/1001 at 192.168.1.4:1065;transport=tcp [CS_HOLD] [NO_ANSWER]
2008-05-14 22:34:22 [ERR] switch_ivr_originate.c:813 switch_ivr_originate()
Cannot create outgoing channel of type [user] cause: [ORIGINATOR_CANCEL]
2008-05-14 22:34:22 [INFO] mod_dptools.c:1551 audio_bridge_function()
Originate Failed.  Cause: ORIGINATOR_CANCEL
2008-05-14 22:34:22 [NOTICE] mod_dptools.c:438 answer_function() Channel
[sofia/default/1002 at 192.168.1.2] has been answered
2008-05-14 22:34:22 [NOTICE] switch_core_session.c:748
switch_core_session_thread() Session 4
(sofia/default/1001 at 192.168.1.4:1065;transport=tcp) Ended
2008-05-14 22:34:22 [NOTICE] switch_core_session.c:750
switch_core_session_thread() Close Channel
sofia/default/1001 at 192.168.1.4:1065;transport=tcp [CS_HANGUP]
2008-05-14 22:34:22 [NOTICE] sofia.c:1992 sofia_handle_sip_i_state() Hangup
sofia/default/1002 at 192.168.1.2 [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE]
2008-05-14 22:34:22 [NOTICE] switch_core_session.c:748
switch_core_session_thread() Session 3 (sofia/default/1002 at 192.168.1.2)
Ended
2008-05-14 22:34:22 [NOTICE] switch_core_session.c:750
switch_core_session_thread() Close Channel sofia/default/1002 at 192.168.1.2
[CS_HANGUP]

#######################################


Thanks for help
Chris






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