Should I create a bug issue to GitHub about this?  I looked at
rr/record.c, but could not figure out, what needs to be changed.

-- Juha

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

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