[Freeswitch-users] Problem with TLS (call setup)
Brian West
brian at freeswitch.org
Wed May 14 14:10:32 PDT 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 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
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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 at freeswitch.org
More information about the FreeSWITCH-users
mailing list