I am using SIP over WebSockets on port 443, most things work correctly, but when I terminate a call that is ringing I can't because port and transport are wrong. Incoming WSS port 443: recv 461 bytes from wss/[50.246.121.173]:34738 at 17:03:00.035305: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/WSS 18.211.78.106:443;rport;branch=z9hG4bKvNgc7Dy2r4rHS From: "Beth User" <sip:beth@18.211.78.106>;tag=B1564gFygm3jN To: <sip:r6ucrocm@3val7ifjblg6.invalid;transport=ws;received= 50.246.121.173:34738>;tag=aa53u3jvjt CSeq: 10377697 INVITE Call-ID: ff8ac04a-5e46-1238-2787-0251a831415e Supported: outbound User-Agent: nathan@stage.illumy1.com Contact: <sip:r6ucrocm@3val7ifjblg6.invalid;transport=ws> Content-Length: 0 Outgoing WSS port 443: send 760 bytes to wss/[50.246.121.173]:34010 at 17:03:00.061145: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/WSS vg5qpm92h7dm.invalid;branch=z9hG4bK2823219;received=50.246.121.173;rport=34010 From: "beth@stage.illumy1.com" <sip:beth@stage.illumy1.com>;tag=ukhk0sdsd9 To: <sip:nathan@stage.illumy1.com>;tag=9eKN1tDQp2pDe Call-ID: reo4ds005e1qlr4784v8 CSeq: 2 INVITE Contact: <sip:nathan@18.211.78.106:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.10.0-release~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 As you can see, Contact incorrectly states port and transport.
<> nathan stratton