recheck phonerlite Route header processing.
seems, that destination udp port obtaining is incorrect.
example (in-dialog hold re-INVITE from phonerlite. phonerlite is callee):
INVITE sip:1111@192.168.1.200:6050 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK0034b7b3ed0ce711962af7f8f429306e;rport
Route: <sip:192.168.1.222;lr;ftag=as483afbb6;did=79b.c1d1>
From: <sip:test@192.168.1.222:5060>;tag=80168bafed0ce711962af7f8f429306e
To: "murfur" <sip:1111@192.168.1.200:6050>;tag=as483afbb6
Call-ID: 7ad031a579e0ba634ef37ce2554d1415@192.168.1.200:6050
CSeq: 40 INVITE
Contact: <sip:test@192.168.1.101:5060>
Content-Type: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Max-Forwards: 70
Supported: 100rel, replaces, from-change
User-Agent: SIPPER for PhonerLite
Content-Length: 498
v=0
o=- 2721632149 2 IN IP4 192.168.1.101
s=SIPPER for PhonerLite
c=IN IP4 192.168.1.101
t=0 0
m=audio 5062 RTP/AVP 107 8 0 2 3 97 110 111 9 18 101
a=rtpmap:107 opus/48000/2
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:110 speex/8000
a=rtpmap:111 speex/16000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:723172387
a=sendrecv
this message is expected to be sent (according to the Route header) via 192.168.1.222:5060, but actual behavior is outgoing request to the 192.168.1.222:6050.
it looks like transport address is from Route URI domain part, but port is from RURI.
it can be simply reproduced with any configuration when there is SIP-proxy on the path between phonerlite and UA and UA and SIP-proxy are work on the different ports.
(in other words: port in the Contact header from the UA is not equal to the port that proxy is listen for)