Re: [Freeswitch-users] Call UDP -> TLS

27 Mar 2013

      For test replace FreeSWITCH witch Opensips for register.
Phone -> TLS -> Opensips - > UDP -> Opensips
Work fine.

172.20.0.24.5060 > 172.20.0.22.5060: [udp sum ok] SIP, length: 919
    REGISTER sip:172.20.0.22:5060 SIP/2.0
    Via: SIP/2.0/UDP 172.20.0.24;branch=z9hG4bK4529.c9ec9a7.0
    Via: SIP/2.0/UDP 172.20.0.20:5060
;received=172.20.0.20;branch=z9hG4bK512348322;rport=5060
    From: <sip:1001@172.20.0.24>;tag=319008323
    To: <sip:1001@172.20.0.24>
    Call-ID: 1014532685-5060-1@BHC.CA.A.CA
    CSeq: 2011 REGISTER
    Contact: <sip:1001@172.20.0.20:5060
...
;reg-id=1;+sip.instance="<urn:uuid:00000000-0000-1000-8000-000B823DB6B2>"
    Authorization: Digest username="1001", realm="172.20.0.24",
nonce="51533e2a000000218d8794b71cd2825d45bcad006b247f2a",
uri="sip:172.20.0.24", response="b18f3fbbede44642c33564bffce9c1ea",
algorithm=MD5
    Max-Forwards: 30
    User-Agent: Grandstream GXP1405 1.0.5.10
    Supported: path
    Expires: 300
    Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO,
REFER, UPDATE, MESSAGE
    Content-Length: 0
    Path: <sip:172.20.0.24;lr;received=sip:172.20.0.20:5060>
    X-AUTH-IP: 172.20.0.20
172.20.0.22.5060 > 172.20.0.24.5060: [bad udp cksum 16dc!] SIP, length: 517
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 172.20.0.24;branch=z9hG4bK4529.c9ec9a7.0
    Via: SIP/2.0/UDP 172.20.0.20:5060
;received=172.20.0.20;branch=z9hG4bK512348322;rport=5060
    From: <sip:1001@172.20.0.24>;tag=319008323
    To: <sip:1001@172.20.0.24>;tag=64e8f587865022a9970237a3f630ac12.72d7
    Call-ID: 1014532685-5060-1@BHC.CA.A.CA
    CSeq: 2011 REGISTER
    Contact: <sip:1001@172.20.0.20:5060>;expires=300
    Path: <sip:172.20.0.24;lr;received=sip:172.20.0.20:5060>
    Server: OpenSIPS (1.9.0-notls (x86_64/linux))
    Content-Length: 0

Call from 1000 to 1001

172.20.0.24.5060 > 172.20.0.22.5060: [udp sum ok] SIP, length: 1223
    INVITE sip:1000@172.20.0.22:5060 SIP/2.0
    Record-Route: <sip:172.20.0.24;lr;ftag=256750377;did=a69.7eedcea5>
    Via: SIP/2.0/UDP 172.20.0.24;branch=z9hG4bK9cef.fef718b6.0
    Via: SIP/2.0/UDP 172.20.0.20:5060
;received=172.20.0.20;branch=z9hG4bK69130358;rport=5060
    From: "1001" <sip:1001@172.20.0.24>;tag=256750377
    To: <sip:1000@172.20.0.24>
    Call-ID: 1487412933-5060-4@BHC.CA.A.CA
    CSeq: 30 INVITE
    Contact: "1001" <sip:1001@172.20.0.20:5060>
    Max-Forwards: 30
    User-Agent: Grandstream GXP1405 1.0.5.10
    Privacy: none
    P-Preferred-Identity: "1001" <sip:1001@172.20.0.24>
    Supported: replaces, path, timer
    Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO,
REFER, UPDATE, MESSAGE
    Content-Type: application/sdp
    Accept: application/sdp, application/dtmf-relay
    Content-Length:   400
    X-AUTH-IP: 172.20.0.20

    v=0
    o=1001 8000 8000 IN IP4 172.20.0.20
    s=SIP Call
    c=IN IP4 172.20.0.20
    t=0 0
    m=audio 5004 RTP/AVP 0 8 4 18 9 97 2 101
    a=sendrecv
    a=rtpmap:0 PCMU/8000
    a=ptime:20
    a=rtpmap:8 PCMA/8000
    a=rtpmap:4 G723/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:9 G722/8000
    a=rtpmap:97 iLBC/8000
    a=fmtp:97 mode=30
    a=rtpmap:2 G726-32/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15

172.20.0.22.5060 > 172.20.0.24.5060: [bad udp cksum bb07!] SIP, length: 380
    SIP/2.0 100 Giving a try
    Via: SIP/2.0/UDP 172.20.0.24;branch=z9hG4bK9cef.fef718b6.0
    Via: SIP/2.0/UDP 172.20.0.20:5060
;received=172.20.0.20;branch=z9hG4bK69130358;rport=5060
    From: "1001" <sip:1001@172.20.0.24>;tag=256750377
    To: <sip:1000@172.20.0.24>
    Call-ID: 1487412933-5060-4@BHC.CA.A.CA
    CSeq: 30 INVITE
    Server: OpenSIPS (1.9.0-notls (x86_64/linux))
    Content-Length: 0

172.20.0.22.5060 > 172.20.0.24.5060: SIP, length: 1472
    INVITE sip:1000@172.20.0.16:5060;transport=tls;transport=tls SIP/2.0
    Record-Route: <sip:172.20.0.22;lr;did=a69.1fbc0244>
    Record-Route: <sip:172.20.0.24;lr;ftag=256750377;did=a69.7eedcea5>
    Via: SIP/2.0/UDP 172.20.0.22:5060;branch=z9hG4bK9cef.1674f7f1.0
    Via: SIP/2.0/UDP 172.20.0.24;branch=z9hG4bK9cef.fef718b6.0
    Via: SIP/2.0/UDP 172.20.0.20:5060
;received=172.20.0.20;branch=z9hG4bK69130358;rport=5060
    Route: <sip:172.20.0.24;r2=on;lr;received=sip:172.20.0.16:36572
...
,<sip:172.20.0.24:5061;transport=tls;r2=on;lr>
    From: "1001" <sip:1001@172.20.0.24>;tag=256750377
    To: <sip:1000@172.20.0.24>
    Call-ID: 1487412933-5060-4@BHC.CA.A.CA
    CSeq: 30 INVITE
    Contact: "1001" <sip:1001@172.20.0.20:5060>
    Max-Forwards: 29
    User-Agent: Grandstream GXP1405 1.0.5.10
    Privacy: none
    P-Preferred-Identity: "1001" <sip:1001@172.20.0.24>
    Supported: replaces, path, timer
    Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO,
REFER, UPDATE, MESSAGE
    Content-Type: application/sdp
    Accept: application/sdp, application/dtmf-relay
    Content-Length:   400
    X-AUTH-IP: 172.20.0.20
v=0
    o=1001 8000 8000 IN IP4 172.20.0.20
    s=SIP Call
    c=IN IP4 172.20.0.20
    t=0 0
    m=audio 5004 RTP/AVP 0 8 4 18 9 97 2 101
    a=sendrecv
    a=rtpmap:0 PCMU/8000
    a=ptime:20
    a=rtpmap:8 PCMA/8000
    a=rtpmap:4 G723/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:9 G722/8000
    a=rtpmap:97 iLBC/8000
    a=fmtp:97 mode=30
    a=rtpmap:2 G726-32/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:[|sip]

172.20.0.24.5060 > 172.20.0.22.5060: [udp sum ok] SIP, length: 398
    SIP/2.0 100 Giving a try
    Via: SIP/2.0/UDP 172.20.0.22:5060;branch=z9hG4bK9cef.1674f7f1.0
    Via: SIP/2.0/UDP 172.20.0.24;branch=z9hG4bK9cef.fef718b6.0
    Via: SIP/2.0/UDP 172.20.0.20:5060
;received=172.20.0.20;branch=z9hG4bK69130358;rport=5060
    From: "1001" <sip:1001@172.20.0.24>;tag=256750377
    To: <sip:1000@172.20.0.24>
    Call-ID: 1487412933-5060-4@BHC.CA.A.CA
    CSeq: 30 INVITE
    Content-Length: 0

172.20.0.24.5060 > 172.20.0.22.5060: [udp sum ok] SIP, length: 942
    SIP/2.0 180 Ringing
    Via: SIP/2.0/UDP 172.20.0.22:5060;branch=z9hG4bK9cef.1674f7f1.0
    Via: SIP/2.0/UDP 172.20.0.24;branch=z9hG4bK9cef.fef718b6.0
    Via: SIP/2.0/UDP 172.20.0.20:5060
;received=172.20.0.20;branch=z9hG4bK69130358;rport=5060
    Record-Route: <sip:172.20.0.24:5061
;transport=tls;r2=on;lr;ftag=256750377;did=a69.8eedcea5>
    Record-Route: <sip:172.20.0.24;r2=on;lr;ftag=256750377;did=a69.8eedcea5>
    Record-Route: <sip:172.20.0.22;lr;did=a69.1fbc0244>
    Record-Route: <sip:172.20.0.24;lr;ftag=256750377;did=a69.7eedcea5>
    From: "1001" <sip:1001@172.20.0.24>;tag=256750377
    To: <sip:1000@172.20.0.24>;tag=1572316537
    Call-ID: 1487412933-5060-4@BHC.CA.A.CA
    CSeq: 30 INVITE
    Contact: <sip:1000@172.20.0.16:36572;transport=tls>
    Supported: replaces, path, timer
    User-Agent: Grandstream GXP2120 1.0.5.14
    Allow-Events: talk, hold
    Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO,
REFER, UPDATE, MESSAGE
    Content-Length: 0

172.20.0.22.5060 > 172.20.0.24.5060: [bad udp cksum b82a!] SIP, length: 877
    SIP/2.0 180 Ringing
    Via: SIP/2.0/UDP 172.20.0.24;branch=z9hG4bK9cef.fef718b6.0
    Via: SIP/2.0/UDP 172.20.0.20:5060
;received=172.20.0.20;branch=z9hG4bK69130358;rport=5060
    Record-Route: <sip:172.20.0.24:5061
;transport=tls;r2=on;lr;ftag=256750377;did=a69.8eedcea5>
    Record-Route: <sip:172.20.0.24;r2=on;lr;ftag=256750377;did=a69.8eedcea5>
    Record-Route: <sip:172.20.0.22;lr;did=a69.1fbc0244>
    Record-Route: <sip:172.20.0.24;lr;ftag=256750377;did=a69.7eedcea5>
    From: "1001" <sip:1001@172.20.0.24>;tag=256750377
    To: <sip:1000@172.20.0.24>;tag=1572316537
    Call-ID: 1487412933-5060-4@BHC.CA.A.CA
    CSeq: 30 INVITE
    Contact: <sip:1000@172.20.0.16:36572;transport=tls>
    Supported: replaces, path, timer
    User-Agent: Grandstream GXP2120 1.0.5.14
    Allow-Events: talk, hold
    Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO,
REFER, UPDATE, MESSAGE
    Content-Length: 0

2013/3/26 Chusov Alexsandr <chusov.alexsandr@gmail.com>
...
2013/3/26 Michael Collins <msc@freeswitch.org>
...
showed specifically has "transport=tls
I wrote in the mailing list  Bogan  response:
Hi Alexsandr,
Well, the problem is more complex a bit. I see you configured OpenSIPs
to add PATH header to REGISTER before sending it to FS. The address in
PATH has UDP transport, so FS (if supports PATH) should send the calls
back to OpenSIPS by using the address from PATH (with UDP). The contact
in register has TLS transport (and you fwd it to FS as it is), but it
should not be used directly by FS because of the presence and priority
of PATH hdr.
So, my only explanation is that FS does not support / not configured to
handle PATH, so that it uses the address from contact, which is TLS.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developerhttp://www.opensips-solutions.com