Record route ignoted while sending ack - sip

Question

I have a strange problem where pjsip ignores the record route info while sending the ack. Below are the sip message flow from the logs:
INVITE sip:+110#xxx.com;transport=tls SIP/2.0
Via: SIP/2.0/TLS ipv4.addr:38890;rport;branch=z9hG4bKPjdYP6TZrj4w7v8kicC3cBgABBNb47QHH2;alias
Max-Forwards: 70
From: "+558" <sip:+558#xxx.com>;tag=qfc3TEYcpfIBQHVXMOmh.7pyvqgmVdMh
To: sip:+110#xxx.com
Contact: "+558" <sip:+558#xxx.com>
Call-ID: 7FdLGhQ1L5BjAQsUrCPEOB3WbXipRfs1
CSeq: 18162 INVITE
Route: <sip:xxx.com:5061;transport=tls;lr>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
User-Agent: SecuVOICE BB10 CSE 2.14.0.1 on Z10 10.3.1.2243
Authorization: Digest xxxx
Content-Type: application/x-x509-user-cert
Content-Length:
SIP/2.0 200 OK
Max-Forwards: 10
Via: SIP/2.0/TLS ipv4.addr:38890;rport=38890;received=ipv4.addr;branch=z9hG4bKPjdYP6TZrj4w7v8kicC3cBgABBNb47QHH2;alias
Record-Route:<sip:xxx.com:5061;transport=tls;lr;ftag=qfc3TEYcpfIBQHVXMOmh.7pyvqgmVdMh;cookie_=e43.052768f7>
Call-ID: 7FdLGhQ1L5BjAQsUrCPEOB3WbXipRfs1
From: "+558" <sip:+558#xxx.com>;tag=qfc3TEYcpfIBQHVXMOmh.7pyvqgmVdMh
To: <sip:+110#xxx.com>;tag=RuDb.RX-9YD0V.BKh0rpj61-SK-ORE5B
CSeq: 18162 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Contact: "+110" <sip:+110#ipv4.addr:25365>
Supported: replaces, 100rel, timer, norefersub
Content-Type: multipart/mixed;boundary=SBC1hJLGTAfp3t2j3HYWIvvgUBsC1RpJ
Content-Length: 27
ACK sip:+110#ipv4.addr:25365 SIP/2.0
"+110" <sip:+110#ipv4.addr:25365>
Via: SIP/2.0/TLS ipv4.addr:38890;rport;branch=z9hG4bKPjkp-dUZmmgpXNWrZHe2ykqvrr9CgRvlm2;alias
Max-Forwards: 70
From: "+558" <sip:+558#xxx.com>;tag=qfc3TEYcpfIBQHVXMOmh.7pyvqgmVdMh
To: sip:+110#xxx.com;tag=RuDb.RX-9YD0V.BKh0rpj61-SK-ORE5B
Call-ID: 7FdLGhQ1L5BjAQsUrCPEOB3WbXipRfs1
CSeq: 18162 ACK
Route: <sip:xxx.com:5061;transport=tls;lr;ftag=qfc3TEYcpfIBQHVXMOmh.7pyvqgmVdMh;cookie_=e43.052768f7>
Content-Type: application/sdp
Content-Length: 709
Looking at the record route from 200 OK, I expected the ACK to look like
ACK sip:+110#ipv4.addr:25365;transport=tls;lr SIP/2.0
Why pjsip is ignoring the transport uri parameter?

Answer 1

Received Record-Route are copied as Route in a new outgoing request within the dialog.
The exception is if the Record-Route URI does not carry a ";lr" parameter. This is the backward compatible behaviour with RFC 2543
The Request URI of the outgoing request is set to the received Contact header.
See RFC 3261 Section 12.2.1.1
The UAC uses the remote target and route set to build the
Request-URI and Route header field of the request.
If the route set is empty, the UAC MUST place the remote target URI
into the Request-URI. The UAC MUST NOT add a Route header field to
the request.
If the route set is not empty, and the first URI in the route set
contains the lr parameter (see Section 19.1.1), the UAC MUST place
the remote target URI into the Request-URI and MUST include a Route
header field containing the route set values in order, including all
parameters.
If the route set is not empty, and its first URI does not contain
the lr parameter, the UAC MUST place the first URI from the route
set into the Request-URI, stripping any parameters that are not
allowed in a Request-URI. The UAC MUST add a Route header field
containing the remainder of the route set values in order,
including all parameters. The UAC MUST then place the remote
target URI into the Route header field as the last value.
The route set is either pre-configured, or learned through the Record-Route.
Target URI is updated when receiving a Contact header from the other party.