Inbound call without SDP, causes 200 SDP in response to be invalid

[Megamuch]

Initial invite

Session Initiation Protocol (INVITE)
    Request-Line: INVITE sip:[email protected]:5060 SIP/2.0
    Message Header
        Contact: sip:x.x.x.x:5073
        To: <sip:[email protected]>
        From: 6e9f1748-bf09-4ecd-b215-0451745ed048 <sip:[email protected]>;tag=dlv7est5fdaigpqu.o
        Call-ID: faf725caad6a4d83908a1668cc5c0535~o
        [Generated Call-ID: faf725caad6a4d83908a1668cc5c0535~o]
        CSeq: 531 INVITE
        Expires: 300
        Allow: INVITE, ACK, BYE, CANCEL, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS, UPDATE
        P-Asserted-Identity: 6e9f1748-bf09-4ecd-b215-0451745ed048 <sip:[email protected]>
        Content-Length: 0

The initial invite comes in to jbz and is handled without issue, just no SDP is offered in the initial invite.
We get a normal 100 trying response and then a 200 OK when jambonz connects to a demo app (hello world).

The 200 OK response from JBZ is.

Session Initiation Protocol (200)
    Status-Line: SIP/2.0 200 OK
    Message Header
        From: 6e9f1748-bf09-4ecd-b215-0451745ed048 <sip:[email protected]>;tag=dlv7est5fdaigpqu.o
        To: <sip:[email protected]>;tag=KHUNeBQ263v0K
        Call-ID: faf725caad6a4d83908a1668cc5c0535~o
        [Generated Call-ID: faf725caad6a4d83908a1668cc5c0535~o]
        CSeq: 531 INVITE
        Contact: <sip:y.y.y.y:5060>
        Content-Type: application/sdp
        Content-Length: 491
        X-Call-Sid: a39327f6-f7cb-4014-a675-cb935dcXXXXXX
        X-Application-Sid: d0f2ea34-5f53-4adb-a4e1-700556XXXXX
    Message Body
        Session Description Protocol
            Session Description Protocol Version (v): 0
            Owner/Creator, Session Id (o): FreeSWITCH 1691552439 1691552440 IN IP4 10.1.13.222
            Session Name (s): FreeSWITCH
            Connection Information (c): IN IP4 10.1.13.222
            Time Description, active time (t): 0 0
            Media Description, name and address (m): audio 46642 RTP/AVP 0 8 102 9 101 103
            Media Attribute (a): rtpmap:0 PCMU/8000
            Media Attribute (a): rtpmap:8 PCMA/8000
            Media Attribute (a): rtpmap:102 opus/48000/2
            Media Attribute (a): fmtp:102 useinbandfec=1; maxaveragebitrate=30000; maxplaybackrate=48000; ptime=20; minptime=10; maxptime=40
            Media Attribute (a): rtpmap:9 G722/8000
            Media Attribute (a): rtpmap:101 telephone-event/8000
            Media Attribute (a): fmtp:101 0-16
            Media Attribute (a): rtpmap:103 telephone-event/48000
            Media Attribute (a): fmtp:103 0-16
            Media Attribute (a): sendrecv
            Media Attribute (a): rtcp:46643
            Media Attribute (a): ptime:20
            [Generated Call-ID: faf725caad6a4d83908a1668cc5c0535~o]

This appears correct, except the C= line in the SDP is using the local (private) IP instead of the public IP which is what is needed.

I haven't found any other differences when comparing calls with and without initial SDP.