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
> <[email protected]> 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:[email protected]:5061 SIP/2.0
> Via: SIP/2.0/TLS 172.16.1.194:5061;branch=z9hG4bK236557042;rport;alias
> From: <sips:[email protected]>;tag=444048970
> To: <sips:[email protected]>
> Call-ID: [email protected]
> CSeq: 21 INVITE
> Contact: <sips:[email protected]:5061;transport=tls>
> Proxy-Authorization: Digest username="100000000", realm="dev-sip.server.com",
> nonce="Z6Ig0WeiH6Vdxtvej0I3XGCTUAEfh+0/",
> uri="sips:[email protected]",
> 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:[email protected] SIP/2.0
> Via: SIP/2.0/UDP
> 10.64.52.113:5080;branch=z9hG4bK1552.89ec9666294034a5611f908f2a6a427e.0;i=f
> From: "Test" <sip:[email protected]>;tag=444048970
> To: <sip:[email protected]>
> Call-ID: [email protected]
> 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:[email protected]: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.
> __________________________________________________________
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--
Alex Balashov
Principal Consultant
Evariste Systems LLC
Web: https://evaristesys.com
Tel: +1-706-510-6800
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