Hi Alex,
Thank you, that worked. I was over complicating it in my script by doing many
replacements.
On Tuesday, February 4th, 2025 at 7:38 PM, Alex Balashov via sr-users
<sr-users@lists.kamailio.org> wrote:
> Yes, you can switch the schema:
>
> if(proto == TLS)
> $ru = $(ru{s.replace,sips:,sip:});
>
> -- Alex
>
> > On Feb 4, 2025, at 2:35 PM, James Lipski via sr-users
> > sr-users@lists.kamailio.org wrote:
> >
> > Hello,
> >
> > I have a kamailio installation where it's basically doing transport
> > protocol translation. To go into more detail, I have a UA communicating
> > over TLS towards kamailio/ and kamailio forwards the request to a
> > freeswitch installation via UDP, so call-flow is as follows:
> >
> > UA <-- (TLS) --> Kamailio <-- (UDP) --> Freeswitch
> >
> > For the most part, this does work/ it's only when I use a UA that uses
> > "sips:" as the schema instead of "sip:" where issues occur -- in my current
> > topology, freeswitch has some weird behavior where it doesn't disconnect
> > the call properly. At least based on my testing, I believe the issue is
> > relating to the schema -- as the UA in question does have the ability to
> > switch between schemas/ when using "sip:" instead of "sips:" over TLS, no
> > issues with the same call-flow. In the event that I do come across a UA
> > that doesn't have that ability, I was wondering if kamailio has the ability
> > to switch schemas when it sends traffic over the UDP connection.
> >
> > I'm able to replace most aspects of the packet with "sip:" however the only
> > header which contains the "sips:" schema is the contact header. With this
> > alone, it propagates to the B-leg side (UDP side) of the call; and will
> > cause freeswitch to not disconnect calls properly.
> >
> > I'm using the topos module to obfuscate my topology, not sure if that
> > matters.
> > --------
> >
> > For example, on the A-leg side of the call, kamailio receives this:
> > INVITE sips:12125551...@dev-sip.server.com:5061 SIP/2.0
> > Via: SIP/2.0/TLS 172.16.1.194:5061;branch=z9hG4bK236557042;rport;alias
> > From: sips:100000...@dev-sip.server.com;tag=444048970
> > To: sips:12125551...@dev-sip.server.com
> > Call-ID: 1326419916-506...@bhc.bg.b.bje
> > CSeq: 21 INVITE
> > Contact: sips:100000000@172.16.1.194:5061;transport=tls
> > Proxy-Authorization: Digest username="100000000",
> > realm="dev-sip.server.com", nonce="Z6Ig0WeiH6Vdxtvej0I3XGCTUAEfh+0/",
> > uri="sips:12125551...@dev-sip.server.com",
> > response="43c1487d077db36a70fffdd1e25b3b52", algorithm=MD5
> > Max-Forwards: 70
> > User-Agent: Grandstream HT812 1.0.57.1
> > Supported: replaces, path, timer, eventlist
> > Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER,
> > UPDATE
> > Content-Type: application/sdp
> > Accept: application/sdp, application/dtmf-relay
> > Content-Length: 1034
> >
> > v=0
> > o=100000000 8000 8000 IN IP4 172.16.1.194
> > s=SIP Call
> > c=IN IP4 172.16.1.194
> > t=0 0
> > m=audio 5004 RTP/SAVP 0 101
> > a=sendrecv
> > a=rtpmap:0 PCMU/8000
> > a=ptime:20
> > a=rtpmap:101 telephone-event/8000
> > a=crypto:1 AEAD_AES_256_GCM
> > inline:UlX3PAjNBP/aQwtwyf5ymsfJciB+VdFxMElO5zLVooUqmcEzZ8YyQQk9stI=|2^32
> >
> > ------------------------------------------------------
> >
> > On the B-leg side of the call, kamailio forwards the following to
> > freeswitch:
> >
> > INVITE sip:12125551...@fs.server.com SIP/2.0
> > Via: SIP/2.0/UDP
> > 10.64.52.113:5080;branch=z9hG4bK1552.89ec9666294034a5611f908f2a6a427e.0;i=f
> > From: "Test" sip:16461234...@dev-sip.server.com;tag=444048970
> > To: sip:12125551...@fs.server.com
> > Call-ID: 1326419916-506...@bhc.bg.b.bje
> > CSeq: 21 INVITE
> > Max-Forwards: 15
> > Supported: replaces, path, timer, eventlist
> > Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER,
> > UPDATE
> > Content-Type: application/sdp
> > Accept: application/sdp, application/dtmf-relay
> > Content-Length: 316
> > Contact: sips:btpsh-67a21ca2-2dbe86-1@10.64.52.113:5080
> >
> > v=0
> > o=100000000 8000 8000 IN IP4 10.64.52.113
> > s=SIP Call
> > c=IN IP4 10.64.52.113
> > t=0 0
> > m=audio 13060 RTP/AVP 0 101
> > a=rtpmap:0 PCMU/8000
> > a=rtpmap:101 telephone-event/8000
> > a=sendrecv
> > a=ptime:20
> >
> > --------------------------------------------------
> >
> > As with the above, I was wondering if there is a way within kamailio to
> > switch schemas on the b-leg side of this call example; specifically is
> > there a way to update the contact header so it doesn't use the "sips:"
> > schema when it traverses the UDP connection.
> >
> > Thank you for you time.
> > __________________________________________________________
> > Kamailio - Users Mailing List - Non Commercial Discussions --
> > sr-users@lists.kamailio.org
> > To unsubscribe send an email to sr-users-le...@lists.kamailio.org
> > Important: keep the mailing list in the recipients, do not reply only to
> > the sender!
>
>
> --
> Alex Balashov
> Principal Consultant
> Evariste Systems LLC
> Web: https://evaristesys.com
> Tel: +1-706-510-6800
>
> __________________________________________________________
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