[Freeswitch-users] Call UDP -> TLS

[Michael Collins]

I don't know much about OpenSIPS but it looks to me like OpenSIPS is
specifically requesting to use TLS as the transport. The Contact header you
showed specifically has "transport=tls". I know that OpenSIPS is quite a
versatile proxy so it would not surprise me if it is able to do what you
are wanting. I would ask on the OpenSIPS mailing list for tips on how to
accomplish this. Bogan (creator of OpenSIPS) knows a lot about FreeSWITCH
as well. If this can be done they'll be happy to show you how. If you
figure it out please come back and tell us if you had to make changes to
OpenSIPS config, FreeSWITCH config, or both.

Thanks,
Michael

On Mon, Mar 25, 2013 at 4:03 PM, Chusov Alexsander <
chusov.alexsandr at gmail.com> wrote:

>      Hello all,
>
> I'm trying to deploy FreeSWITCH as a back-end for Opensips (
> http://wiki.freeswitch.org/wiki/Opensips ). TLS -> Opensips -> UDP ->
> FreeSWITCH
> TLS work fine end point is registered. But when call phone FreeSWITCH
> send invite use TLS instead of UDP.
>
> Register exsample:
> 172.20.0.24 - opensips
> 172.20.0.22 - freeswitch
> 172.20.0.20 - phone
>
>     REGISTER sip:172.20.0.22:5060 SIP/2.0
>     Via: SIP/2.0/UDP 172.20.0.24;branch=z9hG4bK3641.9fabd823.0;i=15
>     Via: SIP/2.0/TLS
> 172.20.0.20:5060
> ;received=172.20.0.20;branch=z9hG4bK1590064540;rport=47050;alias
>     From: <sip:1001 at 172.20.0.24:5061>;tag=1518243149
>     To: <sip:1001 at 172.20.0.24:5061>
>     Call-ID: 1616382919-5060-1 at BHC.CA.A.CA
>     CSeq: 2505 REGISTER
>     Contact:
> <sip:1001 at 172.20.0.20:5060
> ;transport=tls>;reg-id=1;+sip.instance="<urn:uuid:00000000-0000-1000-8000-000B823DB6B2>"
>     Authorization: Digest username="1001", realm="172.20.0.24",
> nonce="102dacd4-959f-11e2-8317-67a135a6f66b",
> uri="sip:172.20.0.24:5061", response="8f80bc7f8fb5fe5a895a5b39f9b5cde6",
> algorithm=MD5, cnonce="12386720", qop=auth, nc=00000005
>     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;r2=on;lr;received=sip:172.20.0.20:47050>
>     X-AUTH-IP: 172.20.0.20
>
> Call-ID:        1616382919-5060-1 at BHC.CA.A.CA
> User:           1001 at 172.20.0.22
> Contact:        "user"
> <sip:1001 at 172.20.0.20:5060
> ;transport=tls;fs_path=%3Csip%3A172.20.0.24%3Br2%3Don%3Blr%3Breceived%3Dsip%3A172.20.0.20%3A47050%3E>
> Agent:          Grandstream GXP1405 1.0.5.10
> Status:         Registered(TLS)(unknown) EXP(2013-03-26 01:01:18)
> EXPSECS(309)
> Host:           172.20.0.22
> IP:             172.20.0.24
> Port:           5060
> Auth-User:      1001
> Auth-Realm:     172.20.0.24
> MWI-Account:    1001 at 172.20.0.22
>
>
> sofia/internal/sip:1001 at 172.20.0.20:5060
> ;transport=tls;fs_path=%3Csip%3A172.20.0.24%3Br2%3Don%3Blr%3Breceived%3Dsip%3A172.20.0.20%3A47050%3E
>
> I'm not familiar with C + + to test made small changes I understand it's
> not right but it works
> in src/mod/endpoints/mod_sofia/sofia_glue.c  1233, 1348  change
>
>          } else if (!strncasecmp(str, "tls", 3)) {
>                  return SOFIA_TRANSPORT_TCP_TLS;
>          }
>
> to
>          } else if (!strncasecmp(str, "tls", 3)) {
>                  return SOFIA_TRANSPORT_UDP;
>          }
>
>
>
> Can anyone tell how to configure FreeSWITCH for normal UDP->TLS TLS->UDP
> work. Or maybe I'm doing something wrong?
>
>
> Sorry for my english
>
>
>
>
> _________________________________________________________________________
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> http://www.freeswitchsolutions.com
>
> 
> 
>
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-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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