[SR-Users] record_route() from websocket UA to tls UA

[Juha Heinanen]

I noticed that when jsSIP UA that has registered over wss calls another
SIP UA that has registered over tls, record_route() adds only one Route
URI to outgoing INVITE (example below).  This causes BYE to fail.

This issue may be caused by the fact that both UAs register over the
same Kamailio tls listening socket.  Still two Route URIs should be
added in the same way as is done when one UA registers over tcp and the
other over udp.

-- Juha

---------------------------------------------------------------------

09:25:43.639833 wss:87.95.73.155:19458 wss:192.27.134.1:5061

INVITE sip:foo at test.tutpro.com SIP/2.0
Via: SIP/2.0/WSS jovobf9n4svm.invalid;branch=z9hG4bK2639638
Max-Forwards: 69
To: <sip:foo at test.tutpro.com>
From: " Test" <sip:test at test.tutpro.com>;tag=f7lpsuo7cb
Call-ID: ovhd9fu1fl1a66nec011
CSeq: 1387 INVITE
Contact: <sip:test at test.tutpro.com;gr=urn:uuid:e7f92a54-2295-4772-abc8-504be07e94c5>
...

09:25:43.649462 tls:192.27.134.1:5061 tls:87.95.73.155:19461

INVITE sip:foo-0x793ee87a90 at 10.158.141.103:38378;transport=tls SIP/2.0
Record-Route: <sip:192.27.134.1:5061;transport=ws;sn=ext_tls;lr;n2;dtlsf=avp;pm=0;ice>
Via: SIP/2.0/TLS 192.27.134.1:5061;branch=z9hG4bK637c.7e15ef5b84ea054cfc8ea3d4e860521f.0
Via: SIP/2.0/WSS jovobf9n4svm.invalid;rport=19458;received=87.95.73.155;branch=z9hG4bK2639638
Max-Forwards: 68
To: <sip:foo at test.tutpro.com>
From: " Test" <sip:test at test.tutpro.com>;tag=f7lpsuo7cb
Call-ID: ovhd9fu1fl1a66nec011
CSeq: 1387 INVITE
Contact: <sip:test at test.tutpro.com;gr=urn:uuid:e7f92a54-2295-4772-abc8-504be07e94c5>