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

14 May 2008


      I think minisip needs to make a choice here.. does it want TLS or does  
it want TCP.. if you notice on the invite the transport=tcp.

/b

On May 14, 2008, at 4:05 PM, Krzysiek wrote:
...
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@trixswitch03> recv 669 bytes from tls/[192.168.1.3]:1072 at
20:33:52.506179:
------------------------------------------------------------------------
  INVITE sip:1001@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@192.168.1.2>;tag=30769
  To: <sip:1001@192.168.1.2>
  Call-ID: 5206@192.168.1.3
  CSeq: 501 INVITE
  Max-Forwards: 70
  Contact: <sip:1002@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@192.168.1.2>;tag=30769
  To: <sip:1001@192.168.1.2>
  Call-ID: 5206@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@192.168.1.2>;tag=30769
  To: <sip:1001@192.168.1.2>;tag=127yN23jBU4HF
  Call-ID: 5206@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@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@192.168.1.2>;tag=30769
  Call-ID: 5206@192.168.1.3
  CSeq: 501 ACK
  To: <sip:1001@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@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@192.168.1.2>;tag=30769
  To: <sip:1001@192.168.1.2>
  Call-ID: 5206@192.168.1.3
  CSeq: 502 INVITE
  Max-Forwards: 70
  Contact: <sip:1002@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@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@192.168.1.2>;tag=30769
  To: <sip:1001@192.168.1.2>
  Call-ID: 5206@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@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@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@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@192.168.1.2>;tag=30769
  To: <sip:1001@192.168.1.2>;tag=2B1QQXmp83t4a
  Call-ID: 5206@192.168.1.3
  CSeq: 502 INVITE
  Contact: <sip:mod_sofia@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@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@192.168.1.2>;tag=30769
  Call-ID: 5206@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@192.168.1.2 
",response="cdf7016d5fe00d8c715047a9edc094e8"
  To: <sip:1001@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@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@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@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@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@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@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@192.168.1.2
[CS_HANGUP]
#######################################
Thanks for help
Chris
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
Brian West
sip:brian@freeswitch.org