<div dir="ltr">Hi Nathan,<div><br></div><div>If you could provide the full SIP trace for a sample, that would be great.</div><div><br></div><div>While the Contact header is indeed typically pointing back at the standard SIP IP/port combination (i.e. non-WebSocket), most JS SIP client library gracefully work around it (even without rport hints).</div><div><br></div><div>Do you use a public stack for the client portion or a implementation?</div><div><br></div><div>-Ciprian</div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Mon, 30 Sep 2019 at 13:46, Nathan Stratton <<a href="mailto:nathan@robotics.net">nathan@robotics.net</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div dir="ltr"><div>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.</div><div><br></div><div>Incoming WSS port 443:</div><div><br></div><div>recv 461 bytes from wss/[50.246.121.173]:34738 at 17:03:00.035305:<br>------------------------------------------------------------------------<br>SIP/2.0 180 Ringing<br>Via: SIP/2.0/WSS 18.211.78.106:443;rport;branch=z9hG4bKvNgc7Dy2r4rHS<br>From: "Beth User" <<a href="mailto:sip%3Abeth@18.211.78.106" target="_blank">sip:beth@18.211.78.106</a>>;tag=B1564gFygm3jN<br>To: <sip:r6ucrocm@3val7ifjblg6.invalid;transport=ws;received=<a href="http://50.246.121.173:34738" target="_blank">50.246.121.173:34738</a>>;tag=aa53u3jvjt<br>CSeq: 10377697 INVITE<br>Call-ID: ff8ac04a-5e46-1238-2787-0251a831415e<br>Supported: outbound<br>User-Agent: <a href="mailto:nathan@stage.illumy1.com" target="_blank">nathan@stage.illumy1.com</a><br>Contact: <sip:r6ucrocm@3val7ifjblg6.invalid;transport=ws><br>Content-Length: 0<br></div><div><br></div><div>Outgoing WSS port 443:</div><div><br></div><div>send 760 bytes to wss/[50.246.121.173]:34010 at 17:03:00.061145:<br>------------------------------------------------------------------------<br>SIP/2.0 180 Ringing<br>Via: SIP/2.0/WSS vg5qpm92h7dm.invalid;branch=z9hG4bK2823219;received=50.246.121.173;rport=34010<br>From: "<a href="mailto:beth@stage.illumy1.com" target="_blank">beth@stage.illumy1.com</a>" <<a href="mailto:sip%3Abeth@stage.illumy1.com" target="_blank">sip:beth@stage.illumy1.com</a>>;tag=ukhk0sdsd9<br>To: <<a href="mailto:sip%3Anathan@stage.illumy1.com" target="_blank">sip:nathan@stage.illumy1.com</a>>;tag=9eKN1tDQp2pDe<br>Call-ID: reo4ds005e1qlr4784v8<br>CSeq: 2 INVITE<br>Contact: <sip:nathan@18.211.78.106:5060;transport=udp><br>User-Agent: FreeSWITCH-mod_sofia/1.10.0-release~64bit<br>Accept: application/sdp<br>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE<br>Supported: path, replaces<br>Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer<br>Content-Length: 0<br></div><div><div dir="ltr"><div dir="ltr"><div><br></div><div>As you can see, Contact incorrectly states port and transport.</div><div><br>><><br>nathan stratton</div></div></div></div></div>
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<a href="https://freeswitch.com" rel="noreferrer" target="_blank">https://freeswitch.com</a></blockquote></div><br clear="all"><div><br></div>-- <br><div dir="ltr" class="gmail_signature">Best Regards,<br>Ciprian Dosoftei<br><br>The information transmitted is intended only for the addressee and may contain privileged and/or confidential material. If you are not the intended recipient, kindly contact the sender and delete the message.<br><br>Any disclosure, distribution or copying of this message is strictly prohibited without the expressed permission of the sender.</div>